From chandranraviram at gmail.com Sun Oct 1 07:42:58 2017 From: chandranraviram at gmail.com (Raviram Chandran) Date: Sun, 1 Oct 2017 13:12:58 +0530 Subject: [Freeswitch-users] How to overwrite Q850 errors code in freeswitch? In-Reply-To: References: Message-ID: Since we have a errors code in PSTN call (IP -PSTN call) which we wanton our telephony server to pass those errors code directly to mobile apps. Presently error codes are configured at network provider end which we need to pass from our telephony server to mobile apps Any help will be appreciated On Thu, Sep 28, 2017 at 12:32 PM, Raviram Chandran < chandranraviram at gmail.com> wrote: > Since we have a errors code in PSTN call (IP -PSTN call) which we wanton > our telephony server to pass those errors code directly to mobile apps. > > When subscriber's balance is lower than some predefined value during call > setup, Warning header with SIP 180/183 response for low balance alert goes > to calling Party (mobile client) and the call continues. > > If heart-beat with calling dialog fails or maximum time limit for the call > is reached, Warning header with BYE message goes to calling party (mobile > client) and the call is dropped. > > Other than this, following failure scenarios may hit during a call > > 1. Authentication failed > 2. Account temporarily deactivated > 3. Zero or inadequate balance while making call > 4. Account already in use by configured number of users > 5. Rate not defined > 6. Called destination blacklisted > 7. Any other application failure For all such failure scenarios, IN > will send 408 SIP responses to client. This response will contain a Warning > header specifying cause of failure and appropriate text message > (configurable at IN). > > Format of Warning Header is as follows > > Warning: Cause_Code APP_Name "Error Text", e.g., > > Presently error codes are configured at network provider end which we need > to pass from our telephony server to mobile apps > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Sun Oct 1 21:52:00 2017 From: nneul at mst.edu (Nathan Neulinger) Date: Sun, 1 Oct 2017 16:52:00 -0500 Subject: [Freeswitch-users] RRNoise - would be pretty cool to see a plugin for this in FS such as for the conference module Message-ID: https://tech.slashdot.org/story/17/10/01/2014216/donate-you-noise-to-xiphmozillas-deep-learning-noise-suppression-project https://people.xiph.org/~jm/demo/rnnoise/ -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From shaun.stokes at itec-support.co.uk Mon Oct 2 06:58:05 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 2 Oct 2017 06:58:05 +0000 Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <030601d33865$159699c0$40c3cd40$@convergedgroup.net> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From shaun.stokes at itec-support.co.uk Mon Oct 2 07:09:36 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 2 Oct 2017 07:09:36 +0000 Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk> Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From mirkobrankovic at gmail.com Mon Oct 2 07:26:08 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Mon, 2 Oct 2017 09:26:08 +0200 Subject: [Freeswitch-users] ICE/DTLS handshake In-Reply-To: References: Message-ID: And I was wrong, nothnig to do with STUN, Problem is that engine state (switch_rtp_ready(engine->rtp_session) is false) is not ready at the moment that STUN/ICE is starting. Not to see what is holding Audio engine to get to ready state (I'm suspecting the TRANSCODING_NECESSARY event) On Thu, Sep 28, 2017 at 3:46 PM, Mirko Brankovic wrote: > actually the answering leg responds with CONTROLING request > > On Thu, Sep 28, 2017 at 3:33 PM, Mirko Brankovic > wrote: > >> Looks like the problem lies in fact that both call legs/channels in refer >> call scenario are in my case outgoing legs, and then they are sending >> CONTROLED stun username requests, so I don't have CONTROLING side. >> >> while at the same time Video stun is negotiated immediately and correctly. >> Log shows it I guess. >> https://pastebin.freeswitch.org/view/33d30c7e >> >> On Sat, Sep 23, 2017 at 3:12 PM, Mirko Brankovic < >> mirkobrankovic at gmail.com> wrote: >> >>> Thanks Mike, >>> all clients are webrtc clients behind same freeswitch. >>> At the same time video rtp/rtcp dtls is instant, but audio is waiting >>> for something, my best guess is for rtcp from client to confirm correct >>> ip.port or to do auto correct. >>> Call scenario is A call B and B transfers(refer) call to C. >>> Also I see in those 5 seconds that first client, A sends total of 6 stun >>> username requests and that C answers to them all at the same time, after 5s. >>> Can this be rtcp problem. >>> I was thinking to go through video ice thread and compare it to audio to >>> see how that one works instantly. >>> thanks, >>> Mirko >>> >>> >>> >>> On Sep 21, 2017 19:55, "Michael Jerris" wrote: >>> >>> its not going to negotiate until we get the stun responses. If we are >>> not, you should look if the client is sending them and something is >>> blocking, or why the client is waiting to send them. Sounds broken on >>> client side from the description. >>> >>> On Sep 21, 2017, at 7:53 AM, Mirko Brankovic >>> wrote: >>> >>> HI, >>> Has anyone experienced DTLS handshake takes 5s to get to SETUP state: >>> >>>> 2017-09-21 11:40:40.021178 [INFO] switch_rtp.c:3515 Changing audio DTLS >>>> state from OFF to HANDSHAKE >>>> 2017-09-21 11:40:45.319457 [INFO] switch_rtp.c:3172 Changing audio DTLS >>>> state from HANDSHAKE to SETUP >>> >>> >>> In network dump i see that answering side is not sending STUN for this >>> 5s and then suddenly answers last 5 STUNs from A side. >>> >>> Has anyone encountered this kind of problem ? >>> >>> I have a pcap if necessary... >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >> >> >> -- >> Regards, >> Mirko >> > > > > -- > Regards, > Mirko > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Mon Oct 2 10:02:49 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 2 Oct 2017 10:02:49 +0000 Subject: [Freeswitch-users] Bug - LUA session:recordFile when using video phones In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A1595@mbx-01.sysconfig.co.uk> , Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2B99@mbx-01.sysconfig.co.uk> Looking at 'src/switch_ivr_play_say.c', this is starting to make a bit more sense. The 10 second delay is produced by line 495 in 'src/switch_ivr_play_say.c' which waits for 10 seconds for video params before continuing, line 496 outputs the error we see 'Unable to establish inbound video stream' just before recording starts. What's the longest amount of time it could take to get video params, 10 seconds seems like a long time to wait for a VoIP call? If we're saving using an audio file prefix (wav\mp3) perhaps we can skip video recording altogether? I'll raise a JIRA. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Jerris [mike at jerris.com] Sent: 30 September 2017 01:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Bug - LUA session:recordFile when using video phones This is fixed in 1.6 as well.. it’s jira wasn’t updated, i have on my backlog to go back and fix these in jira to properly reflect the right fix version On Fri, Sep 29, 2017 at 3:50 PM jungle Boogie > wrote: On 28 September 2017 at 05:04, Shaun Stokes > wrote: > Hi All, > > FreeSWITCH version 1.6.19 > > Looks like this is a known issue: > > https://groups.google.com/forum/#!topic/2600hz-users/61XyiXya5Ik > > https://freeswitch.org/jira/si/jira.issueviews:issue-html/FS-9632/FS-9632.html > > > > The JIRA “FS-9632” was marked as resolved but doesn’t appear to apply to LUA > when using “session:recordFile”. > > https://freeswitch.org/jira/browse/FS-9632 > > > > Perhaps we’ve missed something, does anyone have any ideas? > > Well that bug has a fix version of a release that hasn't happened yet, so I don't know if that means it's not in your 1.6.19 build. Here's the commit: https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/efc2ed2a49372e1160b8cdd84872e7876ef01779 At worse, you can make a bug and reference the one you found above. > > Thanks, > > Shaun > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From andrew at convergedgroup.net Mon Oct 2 09:26:52 2017 From: andrew at convergedgroup.net (Andrew Colin) Date: Mon, 2 Oct 2017 11:26:52 +0200 (SAST) Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk> Message-ID: <03be01d33b60$3c0f6f20$b42e4d60$@convergedgroup.net> Where can I get the 3rd party call firmware Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shaun.stokes at itec-support.co.uk Mon Oct 2 11:38:40 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Mon, 2 Oct 2017 11:38:40 +0000 Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <03be01d33b60$3c0f6f20$b42e4d60$@convergedgroup.net> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk>, <03be01d33b60$3c0f6f20$b42e4d60$@convergedgroup.net> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2C11@mbx-01.sysconfig.co.uk> The phone will need to be pre-loaded with the 3PCC firmware by Cisco, if you buy a phone which is pre-loaded with enterprise firmware you can't upgrade to 3PCC, also if you upgrade a 3PCC phone to enterprise you can't switch back it will be stuck on enterprise. The enterprise firmware is designed for CUCM (Cisco proprietary) however is 'supposed' to be backwards compatible with other SIP platforms. If you're already using the enterprise firmware I can see they've recently released version 12 which we haven't tested, perhaps this has better backwards compatibility with other SIP platforms but can't confirm. The 3PCC (MPP) firmware might not be available for all models. You'll need a Cisco account to login but it's free to setup and they don't charge for the firmware: https://software.cisco.com/download/navigator.html ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 02 October 2017 10:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Where can I get the 3rd party call firmware Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From ovoshlook at gmail.com Mon Oct 2 11:46:28 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Mon, 2 Oct 2017 14:46:28 +0300 Subject: [Freeswitch-users] api:execute("chat") how to set different caller name and number Message-ID: Hi from now i using api:execute("chat","sip|"..userNumber.."@mydomain|external/sip:"..toUser.."@mydomain|"..message:getBody()) And it works find but userNumber also sets at the uuerName parameter at the from field: from: "1001" I wanna control username here and send it as: from: "Peter Pen" Could you help me to find a way hoe to do it? didnt found any informaion about it at the documentation. -------------- next part -------------- An HTML attachment was scrubbed... URL: From nneul at mst.edu Mon Oct 2 14:49:59 2017 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 2 Oct 2017 09:49:59 -0500 Subject: [Freeswitch-users] Any way to configure failover for shoutcast moh streams? Message-ID: I know with the plain file MOH you can point to a directory and have it shuffle - is there any way to do that with streams or to have it fail over from a stream to a static file? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From dig1234 at gmail.com Mon Oct 2 20:43:00 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Mon, 2 Oct 2017 16:43:00 -0400 Subject: [Freeswitch-users] Temporary Equivalent of fs_path In-Reply-To: References: Message-ID: I think you are looking for: sip_invite_route_uri On Mon, Sep 25, 2017 at 11:10 AM, Colin Morelli wrote: > Hey all, > > Trying to figure out how to get the equivalent behavior of fs_path, but > only for a single transaction. In other words, I want to start a SIP > request hitting a particular proxy, and then simply let Record-Route > headers determine where the call should be routed after that. > > I might be completely overthinking this, but everything I've tried so far > (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the > given route for all subsequent requests, rather than simply falling back to > the session route. > > Am I missing something? > > Best, > Colin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at convergedgroup.net Mon Oct 2 11:31:46 2017 From: andrew at convergedgroup.net (Andrew Colin) Date: Mon, 2 Oct 2017 13:31:46 +0200 (SAST) Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk> Message-ID: <041601d33b71$ae957510$0bc05f30$@convergedgroup.net> Also any chance you can share your config file with me as I cant seem to even get the phone to try register? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From andrew at convergedgroup.net Mon Oct 2 12:07:34 2017 From: andrew at convergedgroup.net (Andrew Colin) Date: Mon, 2 Oct 2017 14:07:34 +0200 (SAST) Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2C11@mbx-01.sysconfig.co.uk> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk>, <03be01d33b60$3c0f6f20$b42e4d60$@convergedgroup.net> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2C11@mbx-01.sysconfig.co.uk> Message-ID: <042c01d33b76$aef6dcb0$0ce49610$@convergedgroup.net> I have checked there but don’t see a ccp version for the 8845 As far as I can see the phone is using enterprise firmware at the moment Any ideas :( Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 1:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 The phone will need to be pre-loaded with the 3PCC firmware by Cisco, if you buy a phone which is pre-loaded with enterprise firmware you can't upgrade to 3PCC, also if you upgrade a 3PCC phone to enterprise you can't switch back it will be stuck on enterprise. The enterprise firmware is designed for CUCM (Cisco proprietary) however is 'supposed' to be backwards compatible with other SIP platforms. If you're already using the enterprise firmware I can see they've recently released version 12 which we haven't tested, perhaps this has better backwards compatibility with other SIP platforms but can't confirm. The 3PCC (MPP) firmware might not be available for all models. You'll need a Cisco account to login but it's free to setup and they don't charge for the firmware: https://software.cisco.com/download/navigator.html ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 02 October 2017 10:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Where can I get the 3rd party call firmware Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shaun.stokes at itec-support.co.uk Tue Oct 3 11:09:08 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Tue, 3 Oct 2017 11:09:08 +0000 Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <041601d33b71$ae957510$0bc05f30$@convergedgroup.net> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk>, <041601d33b71$ae957510$0bc05f30$@convergedgroup.net> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A3069@mbx-01.sysconfig.co.uk> The configuration for 3PCC uses a similar format to the old Cisco SPAs, seems to work well via HTTPS using a provisioning profile rule. You can download the configuration file from the phone using http://phoneip/admin/cfg.xml The enterprise firmware is very different, we've only tested it via TFTP thus far. The configuration for enterprise is typically produced from the CUCM (Cisco provisioning server). Here's an example config, you'll want to go through each of the parameters and update with the correct settings: SIP admin passwordhere D/M/YA GMT Standard/Daylight Time ntpiphere Unicast 2000 5060 5061 sipserverhere 5060 sipserverhere 5060 true true x-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false 101 3 avt false false 3 labelhere 1 false 10 false 9 labelhere sipserverhere:5060 5060 namehere displaynamehere 2 3 sipexthere sipextpasswordhere false 1 voicemailaccessnumberhere 4 5 sipexthere true false false true 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 5060 16348 20134 184 0 dialplan.xml true 2 sip78xx.11-5-1-18 false false 0 1 0 0 0 1 1 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 en_US 1.0.0.0-1 iso-8859-1 1.0.0.0-1 1 http://example.domain.ext/contacts.xml http://example.domain.ext/services/menu.xml 96 0 96 2 0 3804 false true yourwantiphere false ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 02 October 2017 12:31 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Also any chance you can share your config file with me as I cant seem to even get the phone to try register? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From raman.chv at gmail.com Tue Oct 3 14:29:05 2017 From: raman.chv at gmail.com (Ram) Date: Tue, 3 Oct 2017 19:59:05 +0530 Subject: [Freeswitch-users] Video call issue from webrtc to sip Message-ID: Hi, Configured freeswitch as a media gateway between sip and web clients. When video call initiated from Sip client to web-rtc clients, audio sdp is dropped when responding 200 ok to sip client after getting 200 ok response from webrtc clients. where as video sdp is fine in 200 ok. >From the logs found "marking rejected media" might causing issue. but not getting clue for what is causing issue for rejected media. Enclosed complete log for reference. soa_static.c:1189 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): generating local description soa_static.c:1230 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): upgrade with remote description soa_static.c:1264 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): marking rejected media soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff8d9924960, 0x7ff8b0b7f8e0, ""): called soa_static.c:1446 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): storing local description soa.c:1730 soa_activate() soa_activate(static::0x7ff8b0038c40, (nil)) called soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7ff8b0038c40, [(nil)], [0x7ff8d9926ad0], [0x7ff8d9926acc]) called tport.c:3257 tport_tsend() tport_tsend(0x7ff8b0004930) tpn = UDP/ 5.2.2.6:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 5.2.2.6:506 Regards Raman -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: video-fail.log Type: application/octet-stream Size: 200184 bytes Desc: not available URL: From mathias at celea-consulting.fr Tue Oct 3 16:00:48 2017 From: mathias at celea-consulting.fr (Mathias WOLFF) Date: Tue, 3 Oct 2017 18:00:48 +0200 Subject: [Freeswitch-users] lua and pgpool Message-ID: <1ec91868-dea0-8295-a0a3-c03da6d47273@celea-consulting.fr> Hi, I need to connect my lua script to pgpool to access data form postgresql cluster. For now, i address queries via odbc and lua dbh. But lua dbh is doing also connection pooling as pgpool and for that it is not working. What is the best way ? Thnaks Mathias -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 473 bytes Desc: OpenPGP digital signature URL: From stefan.mititelu92 at gmail.com Wed Oct 4 07:40:27 2017 From: stefan.mititelu92 at gmail.com (Mititelu Stefan) Date: Wed, 4 Oct 2017 10:40:27 +0300 Subject: [Freeswitch-users] mod_conference PIN question Message-ID: Hi guys, I am having some issues on setting up a dialplan dynamic conference with PIN, from the request URI, and all subsequent joins to that conference to be asked for that PIN. Here is what I've done so far [1] and almost works as I want to. The issue I have is: when I try to join an "old" conference, I don't get asked for the PIN that was set when the "new" conference was created. Reading [2], it says "Any later attempt to join the conference must specify the same pin number, if one existed when it was created." (and from this, I understand that the user is asked and must type the same PIN that was set when the conference was created) I've been searching a while and found something about setting "conference_enforce_security=true", but it didn't work for me. Any help on this, greatly appreciated. Thank you, Stefan [1] https://pastebin.com/cfUnjjaw [2] https://freeswitch.org/confluence/display/FREESWITCH/mod_conference#mod_conference-ConferenceDialplanApplication -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Wed Oct 4 10:38:51 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 4 Oct 2017 15:38:51 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] Message-ID: Hi Users, I am using lua script to actually call the curl_sendfile, i am able to successfully POST the file on URL, but i am only curious to know the response variables to get the status. Like i used curl previously and there are two variables that are auto set curl_response_data and curl_response_code, i am looking same in the curl_sendfile. CAn anybody help me in this? P.S: i did tried to send file using curl, but i am not aware that how to do that using the curl -F(--form) option, i can upload a file using commandline linux curl command , but could not mapp the option -F in the freeswitch. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Wed Oct 4 11:06:06 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Wed, 04 Oct 2017 11:06:06 +0000 Subject: [Freeswitch-users] Video calls Message-ID: Dear users, Can we record the video calls in freeswitch. Extension A and extension B having a video call by freeswitch. So can we record it? -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Wed Oct 4 11:12:51 2017 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 4 Oct 2017 13:12:51 +0200 Subject: [Freeswitch-users] Video calls In-Reply-To: References: Message-ID: <0ff56944-6a38-5c16-5a41-ed189a6fc40a@wirelessmundi.com> yes you can, check: https://freeswitch.org/confluence/display/FREESWITCH/Video-recording On 10/04/2017 01:06 PM, Freeswitch user wrote: > Dear users, > > Can we record the video calls in freeswitch. Extension A and extension > B having a video call by freeswitch. So can we record it? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Wed Oct 4 12:38:41 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Wed, 4 Oct 2017 18:08:41 +0530 Subject: [Freeswitch-users] Video calls In-Reply-To: <0ff56944-6a38-5c16-5a41-ed189a6fc40a@wirelessmundi.com> References: <0ff56944-6a38-5c16-5a41-ed189a6fc40a@wirelessmundi.com> Message-ID: Thanks For Reply. I'm trying the same which is mentioned in provided url but getting some error.. DialPlan : Error:- 2017-10-04 09:33:22.678400 [ERR] mod_fsv.c:969 You are asking to write 2048 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 2017-10-04 09:33:22.678400 [ERR] switch_ivr_async.c:1169 Error writing /tmp/firstTest.fsv 2017-10-04 09:33:22.938385 [ERR] mod_fsv.c:969 You are asking to write 2048 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format 2017-10-04 09:33:22.938385 [ERR] switch_ivr_async.c:1169 Error writing /tmp/firstTest.fsv 2017-10-04 09:33:23.198409 [ERR] mod_fsv.c:969 You are asking to write 2048 bytes of data which is not supported. Please set enable_file_write_buffering=false to use .fsv format On Wed, Oct 4, 2017 at 4:42 PM, António Silva wrote: > yes you can, check: https://freeswitch.org/confluence/display/FREESWITCH/ > Video-recording > > On 10/04/2017 01:06 PM, Freeswitch user wrote: > > Dear users, > > Can we record the video calls in freeswitch. Extension A and extension B > having a video call by freeswitch. So can we record it? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > Saludos / Regards / Cumprimentos > António Silva > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 stefan.mititelu92 at gmail.com Wed Oct 4 13:42:37 2017 From: stefan.mititelu92 at gmail.com (Mititelu Stefan) Date: Wed, 4 Oct 2017 16:42:37 +0300 Subject: [Freeswitch-users] mod_conference PIN question In-Reply-To: References: Message-ID: Since I couldn't find anything relevant yet I've created pull request [1] to get pin and mpin via mod_conference API. [1] https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1407/overview --- Stefan On Wed, Oct 4, 2017 at 10:40 AM, Mititelu Stefan < stefan.mititelu92 at gmail.com> wrote: > Hi guys, > > I am having some issues on setting up a dialplan dynamic conference with > PIN, from the request URI, and all subsequent joins to that conference to > be asked for that PIN. > > Here is what I've done so far [1] and almost works as I want to. The issue > I have is: when I try to join an "old" conference, I don't get asked for > the PIN that was set when the "new" conference was created. > > Reading [2], it says "Any later attempt to join the conference must > specify the same pin number, if one existed when it was created." (and from > this, I understand that the user is asked and must type the same PIN that > was set when the conference was created) > > I've been searching a while and found something about setting " > conference_enforce_security=true", but it didn't work for me. > > Any help on this, greatly appreciated. > > Thank you, > Stefan > > [1] https://pastebin.com/cfUnjjaw > > [2] https://freeswitch.org/confluence/display/FREESWITCH/ > mod_conference#mod_conference-ConferenceDialplanApplication > -- --- Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Oct 4 17:42:02 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 4 Oct 2017 18:42:02 +0100 Subject: [Freeswitch-users] Setting vars Message-ID: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? And if so, is there a way to get them out as JSON? From samir.doshi at inextrix.com Thu Oct 5 04:46:22 2017 From: samir.doshi at inextrix.com (Samir Doshi) Date: Thu, 5 Oct 2017 10:16:22 +0530 Subject: [Freeswitch-users] Setting vars In-Reply-To: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> Message-ID: Refer below url if that can be helpful : https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols Sent with Mailtrack <#> Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. On Wed, Oct 4, 2017 at 11:12 PM, Rick Jarvis wrote: > I’m hoping to set a variable on a channel, which I can read back in JSON > (ideally) via ‘show channels’. > > Using ‘uuid_setvar’ should I be able to affect any of the variables that > are show in ‘show channels’? > > And if so, is there a way to get them out as JSON? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 vma at vallimamod.org Thu Oct 5 07:50:05 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 5 Oct 2017 09:50:05 +0200 Subject: [Freeswitch-users] Setting vars In-Reply-To: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> Message-ID: <3CA6A0A4-9C5B-4C46-A3C2-18B1A775E281@vallimamod.org> Hi, You can get the channel list as json with 'show channels as json' ;) And of course, if you set any of the listed variables, your custom value will also be available on the corresponding channel. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 4 Oct 2017, at 19:42, Rick Jarvis wrote: > > I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. > > Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? > > And if so, is there a way to get them out as JSON? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rick at magicmail.mooo.com Thu Oct 5 08:53:29 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 5 Oct 2017 09:53:29 +0100 Subject: [Freeswitch-users] Setting vars In-Reply-To: References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> Message-ID: <73BE8143-F14F-42D7-85F3-F560D1A1D409@magicmail.mooo.com> Have to admit, I’ve not played around with the tables before, and the ‘presence’ element of that page confused me as I thought it was relating to SIP presence! I’ve installed sqlite CLI (on Debian) but if I change into /var/lib/freeswitch/db and do a ’sqlite core.db’ I get: Unable to open database "core.db": file is encrypted or is not a database > On 5 Oct 2017, at 05:46, Samir Doshi wrote: > > Refer below url if that can be helpful : > https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols > > > > Sent with Mailtrack > > Best Regards > -- > Samir Doshi > iNextrix Technologies Pvt. Ltd. > http://www.inextrix.com > > > Disclaimer: > The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. > > On Wed, Oct 4, 2017 at 11:12 PM, Rick Jarvis > wrote: > I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. > > Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? > > And if so, is there a way to get them out as JSON? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rick at magicmail.mooo.com Thu Oct 5 08:54:59 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 5 Oct 2017 09:54:59 +0100 Subject: [Freeswitch-users] Setting vars In-Reply-To: <3CA6A0A4-9C5B-4C46-A3C2-18B1A775E281@vallimamod.org> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> <3CA6A0A4-9C5B-4C46-A3C2-18B1A775E281@vallimamod.org> Message-ID: <89C5BD01-6850-48A0-A875-99913AD6CDFC@magicmail.mooo.com> Wow, that’s great, thanks… is there a list of which commands work ‘as json’? For instance, ‘conference list as json’ doesn’t have the same effect? > On 5 Oct 2017, at 08:50, Vallimamod Abdullah wrote: > > Hi, > > You can get the channel list as json with 'show channels as json' ;) > And of course, if you set any of the listed variables, your custom value will also be available on the corresponding channel. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > >> On 4 Oct 2017, at 19:42, Rick Jarvis > wrote: >> >> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. >> >> Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? >> >> And if so, is there a way to get them out as JSON? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 vma at vallimamod.org Thu Oct 5 09:39:03 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 5 Oct 2017 11:39:03 +0200 Subject: [Freeswitch-users] Setting vars In-Reply-To: <89C5BD01-6850-48A0-A875-99913AD6CDFC@magicmail.mooo.com> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> <3CA6A0A4-9C5B-4C46-A3C2-18B1A775E281@vallimamod.org> <89C5BD01-6850-48A0-A875-99913AD6CDFC@magicmail.mooo.com> Message-ID: <0B390E68-774D-4A94-9346-101083860CC7@vallimamod.org> Hi, I am not aware of any such list. I don't remember exactly how I found this specific command but I guess it's from reading the source. There is also an 'as xml' that works well for channel and call list. You can maybe open a feature request on JIRA to add this feature to other commands. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 5 Oct 2017, at 10:54, Rick Jarvis wrote: > > Wow, that’s great, thanks… is there a list of which commands work ‘as json’? For instance, ‘conference list as json’ doesn’t have the same effect? > >> On 5 Oct 2017, at 08:50, Vallimamod Abdullah > wrote: >> >> Hi, >> >> You can get the channel list as json with 'show channels as json' ;) >> And of course, if you set any of the listed variables, your custom value will also be available on the corresponding channel. >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sipsolutions.fr >> . >> >>> On 4 Oct 2017, at 19:42, Rick Jarvis > wrote: >>> >>> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. >>> >>> Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? >>> >>> And if so, is there a way to get them out as JSON? -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Thu Oct 5 10:15:37 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 5 Oct 2017 11:15:37 +0100 Subject: [Freeswitch-users] Setting vars In-Reply-To: <73BE8143-F14F-42D7-85F3-F560D1A1D409@magicmail.mooo.com> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> <73BE8143-F14F-42D7-85F3-F560D1A1D409@magicmail.mooo.com> Message-ID: Ok so for some reason I have to specify the db file without the extension, then it works. Although there are no tables in it anyway, and adding a channels table doesn’t help. I’m guessing I need to be using mysql or similar if I want to be able to change the fields in channels? > On 5 Oct 2017, at 09:53, Rick Jarvis wrote: > > Have to admit, I’ve not played around with the tables before, and the ‘presence’ element of that page confused me as I thought it was relating to SIP presence! > > I’ve installed sqlite CLI (on Debian) but if I change into /var/lib/freeswitch/db and do a ’sqlite core.db’ I get: > > Unable to open database "core.db": file is encrypted or is not a database > > > >> On 5 Oct 2017, at 05:46, Samir Doshi > wrote: >> >> Refer below url if that can be helpful : >> https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols >> >> >> >> Sent with Mailtrack >> >> Best Regards >> -- >> Samir Doshi >> iNextrix Technologies Pvt. Ltd. >> http://www.inextrix.com >> >> >> Disclaimer: >> The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. >> >> On Wed, Oct 4, 2017 at 11:12 PM, Rick Jarvis > wrote: >> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. >> >> Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? >> >> And if so, is there a way to get them out as JSON? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rick at magicmail.mooo.com Thu Oct 5 13:00:07 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 5 Oct 2017 14:00:07 +0100 Subject: [Freeswitch-users] Setting vars In-Reply-To: References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> <73BE8143-F14F-42D7-85F3-F560D1A1D409@magicmail.mooo.com> Message-ID: <4A14DC93-9A6C-4CA6-A5F3-CBF543467ED3@magicmail.mooo.com> Got it working with ODBC and mysql in the end (not sure what the issue was with sqlite, or the dsn-less odbc string, but hey). Although I’ve got the additional fields showing up in ‘show channels’, altering one of these with ‘uuid_setvar’ doesn’t show the current value in the channels table (confirmed with uuid_getvar), only the value initially set. Is there a way of getting ‘show channels’ to update dynamically? I could run a separate database and store everything in there, but I’m hoping to avoid doubling down on everything if possible…? > On 5 Oct 2017, at 11:15, Rick Jarvis wrote: > > Ok so for some reason I have to specify the db file without the extension, then it works. Although there are no tables in it anyway, and adding a channels table doesn’t help. I’m guessing I need to be using mysql or similar if I want to be able to change the fields in channels? > >> On 5 Oct 2017, at 09:53, Rick Jarvis > wrote: >> >> Have to admit, I’ve not played around with the tables before, and the ‘presence’ element of that page confused me as I thought it was relating to SIP presence! >> >> I’ve installed sqlite CLI (on Debian) but if I change into /var/lib/freeswitch/db and do a ’sqlite core.db’ I get: >> >> Unable to open database "core.db": file is encrypted or is not a database >> >> >> >>> On 5 Oct 2017, at 05:46, Samir Doshi > wrote: >>> >>> Refer below url if that can be helpful : >>> https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols >>> >>> >>> >>> Sent with Mailtrack >>> >>> Best Regards >>> -- >>> Samir Doshi >>> iNextrix Technologies Pvt. Ltd. >>> http://www.inextrix.com >>> >>> >>> Disclaimer: >>> The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. >>> >>> On Wed, Oct 4, 2017 at 11:12 PM, Rick Jarvis > wrote: >>> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. >>> >>> Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? >>> >>> And if so, is there a way to get them out as JSON? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 harangozo.laszlo at tct.hu Tue Oct 3 10:21:20 2017 From: harangozo.laszlo at tct.hu (=?UTF-8?B?SGFyYW5nb3rDsywgTMOhc3psw7M=?=) Date: Tue, 3 Oct 2017 12:21:20 +0200 Subject: [Freeswitch-users] PLAYBACK_STOP event not sent trhough ESL for playback if speed has been changed with uuid_fileman Message-ID: Hi guys, Setup: Inbound calls are managed in a separate app through an ESL connection, so the FS dialplan is simple: If a playback operation is started (ESL ExecuteApplication playback) for such a call then both PLAYBACK_START and PLAYBACK_STOP events are sent for the playback application. Except for the case when during play the speed is changed, for example with the following bgapi call: uuid_fileman 8bca0318-a5f5-4170-b82a-d7ab09d018d4 speed:+1 When the speed is changed in such a way, the PLAYBACK_STOP event won't be sent. Environment: Platform: Windows Branch: v1.6 File being played: Issue present with all sound files I have tried The issue happens in a consistent way (all the time when the described scenario happens) Does anyone have a clue, how to get FS to send the PLAYBACK_STOP events? -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at convergedgroup.net Tue Oct 3 14:13:40 2017 From: andrew at convergedgroup.net (Andrew Colin) Date: Tue, 3 Oct 2017 16:13:40 +0200 (SAST) Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A3069@mbx-01.sysconfig.co.uk> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk>, <041601d33b71$ae957510$0bc05f30$@convergedgroup.net> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A3069@mbx-01.sysconfig.co.uk> Message-ID: <036301d33c51$76062c90$621285b0$@convergedgroup.net> Still cant seem to find the 3PCC firmware for this unit Will other versions work? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Tuesday, October 3, 2017 1:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 The configuration for 3PCC uses a similar format to the old Cisco SPAs, seems to work well via HTTPS using a provisioning profile rule. You can download the configuration file from the phone using http://phoneip/admin/cfg.xml The enterprise firmware is very different, we've only tested it via TFTP thus far. The configuration for enterprise is typically produced from the CUCM (Cisco provisioning server). Here's an example config, you'll want to go through each of the parameters and update with the correct settings: SIP admin passwordhere D/M/YA GMT Standard/Daylight Time ntpiphere Unicast 2000 5060 5061 sipserverhere 5060 sipserverhere 5060 true true x-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false 101 3 avt false false 3 labelhere 1 false 10 false 9 labelhere sipserverhere:5060 5060 namehere displaynamehere 2 3 sipexthere sipextpasswordhere false 1 voicemailaccessnumberhere 4 5 sipexthere true false false true 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 5060 16348 20134 184 0 dialplan.xml true 2 sip78xx.11-5-1-18 false false 0 1 0 0 0 1 1 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 en_US 1.0.0.0-1 iso-8859-1 1.0.0.0-1 1 http://example.domain.ext/contacts.xml http://example.domain.ext/services/menu.xml 96 0 96 2 0 3804 false true yourwantiphere false ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 02 October 2017 12:31 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Also any chance you can share your config file with me as I cant seem to even get the phone to try register? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From shaun.stokes at itec-support.co.uk Thu Oct 5 14:03:59 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 5 Oct 2017 14:03:59 +0000 Subject: [Freeswitch-users] Cisco 8845 In-Reply-To: <036301d33c51$76062c90$621285b0$@convergedgroup.net> References: <030601d33865$159699c0$40c3cd40$@convergedgroup.net>, <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A1C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A2A44@mbx-01.sysconfig.co.uk>, <041601d33b71$ae957510$0bc05f30$@convergedgroup.net> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A3069@mbx-01.sysconfig.co.uk>, <036301d33c51$76062c90$621285b0$@convergedgroup.net> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A3D02@mbx-01.sysconfig.co.uk> Our experience of the enterprise firmware on version 11 isn't great, while the phones work there were certain features such as 3 way conference and changing the SIP registrar port which didn't work. We also found an odd issue with audio dropping out on certain calls (mostly international), seems like VAD is active despite disabling this in the config. These were not issues using the 3PCC firmware, version 12 of the enterprise firmware might provide better compatibility but can't confirm. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 03 October 2017 15:13 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Still cant seem to find the 3PCC firmware for this unit Will other versions work? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Tuesday, October 3, 2017 1:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 The configuration for 3PCC uses a similar format to the old Cisco SPAs, seems to work well via HTTPS using a provisioning profile rule. You can download the configuration file from the phone using http://phoneip/admin/cfg.xml The enterprise firmware is very different, we've only tested it via TFTP thus far. The configuration for enterprise is typically produced from the CUCM (Cisco provisioning server). Here's an example config, you'll want to go through each of the parameters and update with the correct settings: SIP admin passwordhere D/M/YA GMT Standard/Daylight Time ntpiphere Unicast 2000 5060 5061 sipserverhere 5060 sipserverhere 5060 true true x-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false 101 3 avt false false 3 labelhere 1 false 10 false 9 labelhere sipserverhere:5060 5060 namehere displaynamehere 2 3 sipexthere sipextpasswordhere false 1 voicemailaccessnumberhere 4 5 sipexthere true false false true 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 20 Menu http://example.domain.ext/services/menu.xml 5060 16348 20134 184 0 dialplan.xml true 2 sip78xx.11-5-1-18 false false 0 1 0 0 0 1 1 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 en_US 1.0.0.0-1 iso-8859-1 1.0.0.0-1 1 http://example.domain.ext/contacts.xml http://example.domain.ext/services/menu.xml 96 0 96 2 0 3804 false true yourwantiphere false ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 02 October 2017 12:31 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Also any chance you can share your config file with me as I cant seem to even get the phone to try register? Andrew Colin Converged Group | Licensed ISP 0258/IECNS/JAN/09 75 Witkoppen Rd, North Riding, Gauteng, 2162 Office: +27 10 591 4600 | Mobile: map | website | company blog | linkedin -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Monday, October 2, 2017 9:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Cisco 8845 Just to add, the 8841 was on the 3rd Party Call Control firmware (11-0-1MPP). We previously had problems such as phones registering to the wrong server port and 3 way conference not working using the enterprise firmware on the 7821 (11-7-1), these problems did not occur on the 3rd Party Call Control firmware (11-0-1MPP). ________________________________________ From: Shaun Stokes Sent: 02 October 2017 07:58 To: FreeSWITCH Users Help Subject: RE: Cisco 8845 Not sure about the 8845, but the 8841 tested and working on FreeSWITCH 1.6 using TCP and TLS. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Andrew Colin [andrew at convergedgroup.net] Sent: 28 September 2017 15:24 To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Cisco 8845 Hi Guys Has anyone got a cisco 8845 IP phone working on freeswitch? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From rick at magicmail.mooo.com Thu Oct 5 14:36:22 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 5 Oct 2017 15:36:22 +0100 Subject: [Freeswitch-users] Setting vars In-Reply-To: <4A14DC93-9A6C-4CA6-A5F3-CBF543467ED3@magicmail.mooo.com> References: <596C86CD-1BB2-41EE-AF43-0EE3B469FB10@magicmail.mooo.com> <73BE8143-F14F-42D7-85F3-F560D1A1D409@magicmail.mooo.com> <4A14DC93-9A6C-4CA6-A5F3-CBF543467ED3@magicmail.mooo.com> Message-ID: <03DD3C62-2AA8-482A-B14E-99739C6DFE3C@magicmail.mooo.com> Ok reading through other responses to similar questions in the mail list archive, it seems that’s not really an option. I guess I could get my app to manually update the channels table by hooking into it? The other option is to monitor ESL which I’m doing anyway, and run my own copy of the data, but I think I might go for the first option, as at least I know that the calls will be correct, and that I haven’t missed an event… > On 5 Oct 2017, at 14:00, Rick Jarvis wrote: > > Got it working with ODBC and mysql in the end (not sure what the issue was with sqlite, or the dsn-less odbc string, but hey). > > Although I’ve got the additional fields showing up in ‘show channels’, altering one of these with ‘uuid_setvar’ doesn’t show the current value in the channels table (confirmed with uuid_getvar), only the value initially set. > > Is there a way of getting ‘show channels’ to update dynamically? I could run a separate database and store everything in there, but I’m hoping to avoid doubling down on everything if possible…? > >> On 5 Oct 2017, at 11:15, Rick Jarvis > wrote: >> >> Ok so for some reason I have to specify the db file without the extension, then it works. Although there are no tables in it anyway, and adding a channels table doesn’t help. I’m guessing I need to be using mysql or similar if I want to be able to change the fields in channels? >> >>> On 5 Oct 2017, at 09:53, Rick Jarvis > wrote: >>> >>> Have to admit, I’ve not played around with the tables before, and the ‘presence’ element of that page confused me as I thought it was relating to SIP presence! >>> >>> I’ve installed sqlite CLI (on Debian) but if I change into /var/lib/freeswitch/db and do a ’sqlite core.db’ I get: >>> >>> Unable to open database "core.db": file is encrypted or is not a database >>> >>> >>> >>>> On 5 Oct 2017, at 05:46, Samir Doshi > wrote: >>>> >>>> Refer below url if that can be helpful : >>>> https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols >>>> >>>> >>>> >>>> Sent with Mailtrack >>>> >>>> Best Regards >>>> -- >>>> Samir Doshi >>>> iNextrix Technologies Pvt. Ltd. >>>> http://www.inextrix.com >>>> >>>> >>>> Disclaimer: >>>> The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. >>>> >>>> On Wed, Oct 4, 2017 at 11:12 PM, Rick Jarvis > wrote: >>>> I’m hoping to set a variable on a channel, which I can read back in JSON (ideally) via ‘show channels’. >>>> >>>> Using ‘uuid_setvar’ should I be able to affect any of the variables that are show in ‘show channels’? >>>> >>>> And if so, is there a way to get them out as JSON? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 colin.morelli at gmail.com Thu Oct 5 15:18:39 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Thu, 5 Oct 2017 11:18:39 -0400 Subject: [Freeswitch-users] Temporary Equivalent of fs_path In-Reply-To: References: Message-ID: I had tried this previously and the call simply didn't connect. Freeswitch responded with a 503, and there wasn't much in the way of logging that was helpful to debug the issue. I'm sure I'm missing something obvious here. I'll see if I can get any more info on it, though. Best, Colin On Mon, Oct 2, 2017 at 4:43 PM, Daniel Greenwald wrote: > I think you are looking for: sip_invite_route_uri > > On Mon, Sep 25, 2017 at 11:10 AM, Colin Morelli > wrote: > >> Hey all, >> >> Trying to figure out how to get the equivalent behavior of fs_path, but >> only for a single transaction. In other words, I want to start a SIP >> request hitting a particular proxy, and then simply let Record-Route >> headers determine where the call should be routed after that. >> >> I might be completely overthinking this, but everything I've tried so far >> (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the >> given route for all subsequent requests, rather than simply falling back to >> the session route. >> >> Am I missing something? >> >> Best, >> Colin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 chad at apartmentlines.com Fri Oct 6 03:06:33 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Thu, 5 Oct 2017 20:06:33 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket Message-ID: I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality? -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Oct 6 03:58:35 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 06 Oct 2017 03:58:35 +0000 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Or, file a similar request with nginx to not do that behavior, On Thu, Oct 5, 2017 at 11:07 PM Chad Phillips wrote: > I've switched to using Nginx to proxy Verto websockets, and have run into > a small snag: by default, if Nginx doesn't read any data from a proxy > backend within 60 seconds, it closes the connection, even for websockets. > > It appears the recommended solution is to have the server send some kind > of regular keepalive. I poked around in mod_verto.c and found a > 'request.keepalive' variable, but I'm unclear how to set that in the > request, and/or if it even accomplishes what I'm wanting. > > I've solved the issue for now by periodically sending a JSON RPC 'echo' > request along the websocket every 50 seconds, and the reply from the server > is enough to keep the connection open. This is fine, but I am curious if > there's a way to do it just from the server side, and if not, if it's worth > it to add a setting to enable that functionality? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Fri Oct 6 04:40:01 2017 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Fri, 6 Oct 2017 11:40:01 +0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Maybe you should check nginx directive keepalive_timeout http://nginx.org/en/docs/http/ngx_http_core_module.html#keepalive_timeout Hope this help On Oct 6, 2017 11:01, "Anthony Minessale" wrote: > Or, file a similar request with nginx to not do that behavior, > > On Thu, Oct 5, 2017 at 11:07 PM Chad Phillips > wrote: > >> I've switched to using Nginx to proxy Verto websockets, and have run into >> a small snag: by default, if Nginx doesn't read any data from a proxy >> backend within 60 seconds, it closes the connection, even for websockets. >> >> It appears the recommended solution is to have the server send some kind >> of regular keepalive. I poked around in mod_verto.c and found a >> 'request.keepalive' variable, but I'm unclear how to set that in the >> request, and/or if it even accomplishes what I'm wanting. >> >> I've solved the issue for now by periodically sending a JSON RPC 'echo' >> request along the websocket every 50 seconds, and the reply from the server >> is enough to keep the connection open. This is fine, but I am curious if >> there's a way to do it just from the server side, and if not, if it's worth >> it to add a setting to enable that functionality? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Oct 6 05:33:08 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Fri, 6 Oct 2017 11:03:08 +0530 Subject: [Freeswitch-users] After sofia recover call was hangup Message-ID: Hello Guys, I built FS HA solution. Both freeswitch point to same database. Now call is running on node1 and i have crash freeswitch node1 and move floting ip to node2. Now when i fire command "sofia recover" getting below error. freeswitch at freeswitch> sofia recover Recovered 2 call(s) After call was hangup and getting NORMAL_TEMPORARY_FAILURE in every case. Call full logs : https://pastebin.com/TXCy4tfR Is there is a way to solve this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jinserk.baik at gmail.com Thu Oct 5 18:26:04 2017 From: jinserk.baik at gmail.com (Jinserk Baik) Date: Thu, 5 Oct 2017 13:26:04 -0500 Subject: [Freeswitch-users] Call is not cleared after hangup Message-ID: Hi all, I'm making a very simple freeswitch dummy module. It's simply make a conference in each incoming call, and hangup after 10 secs for each call. However, when I use switch_channel_hangup(), the conference was terminated but a zombie call record is remaining. I'm using the latest freeswitch 1.8 f26ba42360f15463c5b65d6cb0367b95ba307901 Here is the log. I've replaced the server ip as BB.BB.BB.BB and the client ip as AA.AA.AA.AA. 2017-10-05 13:18:19.508489 [INFO] switch_core.c:2442 FreeSWITCH Version 1.8.0+git~20170814T222640Z~f26ba42360~64bit (git f26ba42 2017-08-14 22:26:40Z 64bit) FreeSWITCH Started Max Sessions [1000] Session Rate [30] SQL [Enabled] 2017-10-05 13:18:19.509110 [INFO] mod_fsmux.c:173 show calls 0 total. 2017-10-05 13:18:19.509136 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. recv 908 bytes from udp/[AA.AA.AA.AA]:5060 at 13:18:20.495288: ------------------------------------------------------------------------ INVITE sip:28 at BB.BB.BB.BB:5062 SIP/2.0 From: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 To: Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 1 INVITE Via: SIP/2.0/UDP AA.AA.AA.AA:5060;branch=z9hG4bK-15d31-55407c5-6920019f Max-Forwards: 70 Supported: 100rel,replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R11.10.7.HA.E Contact: Content-Type: application/sdp Content-Length: 279 v=0 o=- 1507227330 1 IN IP4 AA.AA.AA.AA s=- c=IN IP4 AA.AA.AA.AA t=0 0 m=audio 12828 RTP/AVP 18 0 8 101 a=silenceSupp:off - - - - a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 2017-10-05 13:18:20.488235 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/010001103 at AA.AA.AA.AA:5060 [36e9ea37-5d94-439a-97be-c3d490db12d6] 2017-10-05 13:18:20.488235 [INFO] mod_dialplan_xml.c:637 Processing 010001103 <010001103>->28 in context public send 409 bytes to udp/[AA.AA.AA.AA]:5060 at 13:18:20.496593: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP AA.AA.AA.AA:5060;branch=z9hG4bK-15d31-55407c5-6920019f From: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 To: Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.8.0+git~20170814T222640Z~f26ba42360~64bit Content-Length: 0 ------------------------------------------------------------------------ 2017-10-05 13:18:20.488235 [INFO] switch_core_session.c:2710 Sending early media 2017-10-05 13:18:20.488235 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/010001103 at AA.AA.AA.AA:5060! send 1063 bytes to udp/[AA.AA.AA.AA]:5060 at 13:18:20.500198: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP AA.AA.AA.AA:5060;branch=z9hG4bK-15d31-55407c5-6920019f From: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 To: ;tag=U7c55U2Sva2pS Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 1 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.8.0+git~20170814T222640Z~f26ba42360~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 Remote-Party-ID: "28" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1507203578 1507203579 IN IP4 BB.BB.BB.BB s=FreeSWITCH c=IN IP4 BB.BB.BB.BB t=0 0 m=audio 23922 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ send 1032 bytes to udp/[AA.AA.AA.AA]:5060 at 13:18:20.500315: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP AA.AA.AA.AA:5060;branch=z9hG4bK-15d31-55407c5-6920019f From: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 To: ;tag=U7c55U2Sva2pS Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA 2017-10-05 13:18:20.488235 [NOTICE] mod_conference.c:1835 Channel [sofia/internal/010001103 at AA.AA.AA.AA:5060] has been answered CSeq: 1 INVITE Contact: ;isfocus User-Agent: FreeSWITCH-mod_sofia/1.8.0+git~20170814T222640Z~f26ba42360~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 247 Remote-Party-ID: "28" ;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1507203578 1507203579 IN IP4 BB.BB.BB.BB s=FreeSWITCH c=IN IP4 BB.BB.BB.BB t=0 0 m=audio 23922 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2017-10-05 13:18:20.488235 [NOTICE] mod_fsmux.c:75 event: channel_answer 36e9ea37-5d94-439a-97be-c3d490db12d6 2017-10-05 13:18:20.508222 [INFO] switch_ivr_async.c:215 Digit parser mod_conference: Setting realm to 'conf' 2017-10-05 13:18:20.508222 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:20.508222 [INFO] mod_fsmux.c:173 show calls 0 total. 2017-10-05 13:18:20.508222 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak;0;0;100 2017-10-05 13:18:20.508222 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 1 recv 622 bytes from udp/[AA.AA.AA.AA]:5060 at 13:18:20.514950: ------------------------------------------------------------------------ ACK sip:28 at BB.BB.BB.BB:5062;transport=udp SIP/2.0 From: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 To: ;tag=U7c55U2Sva2pS Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 1 ACK Via: SIP/2.0/UDP AA.AA.AA.AA:5060;branch=z9hG4bK-15d31-55407d9-43fc6205 Max-Forwards: 70 Supported: 100rel,replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R11.10.7.HA.E Contact: Content-Length: 0 ------------------------------------------------------------------------ 2017-10-05 13:18:21.508236 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:21.508236 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:21.508236 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|talking|floor|vid-floor;0;0;100 2017-10-05 13:18:21.508236 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 2 2017-10-05 13:18:22.508247 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:22.508247 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:22.508247 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:22.508247 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 3 2017-10-05 13:18:23.508238 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:23.508238 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:23.508238 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:23.508238 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 4 2017-10-05 13:18:24.508234 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:24.508234 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:24.508234 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:24.508234 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 5 2017-10-05 13:18:25.508238 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:25.508238 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:25.508238 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:25.508238 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 6 2017-10-05 13:18:26.508225 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:26.508225 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:26.508225 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:26.508225 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 7 2017-10-05 13:18:27.508235 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:27.508235 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:27.508235 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:27.508235 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 8 2017-10-05 13:18:28.508234 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:28.508234 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:28.508234 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:28.508234 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 9 2017-10-05 13:18:29.508232 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:29.508232 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:29.508232 [INFO] mod_fsmux.c:178 conference list +OK Conference 36e9ea37-5d94-439a-97be-c3d490db12d6 (1 member rate: 8000 flags: running|answered|bridge_to|dynamic|exit_sound|enter_sound|json_status) 1;sofia/internal/010001103 at AA.AA.AA.AA :5060;36e9ea37-5d94-439a-97be-c3d490db12d6;010001103;010001103;hear|speak|floor|vid-floor;0;0;100 2017-10-05 13:18:29.508232 [NOTICE] mod_fsmux.c:184 call uuid: 36e9ea37-5d94-439a-97be-c3d490db12d6, count: 10 2017-10-05 13:18:29.508232 [NOTICE] mod_fsmux.c:193 Hangup sofia/internal/010001103 at AA.AA.AA.AA:5060 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-05 13:18:29.528232 [INFO] conference_loop.c:1635 Channel leaving conference, cause: NORMAL_CLEARING send 646 bytes to udp/[AA.AA.AA.AA]:5060 at 13:18:29.539020: ------------------------------------------------------------------------ BYE sip:010001103 at AA.AA.AA.AA:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP BB.BB.BB.BB:5062;rport;branch=z9hG4bKUQmt9KNeg9yjN Max-Forwards: 70 From: ;tag=U7c55U2Sva2pS To: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 113277690 BYE User-Agent: FreeSWITCH-mod_sofia/1.8.0+git~20170814T222640Z~f26ba42360~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2017-10-05 13:18:29.528232 [NOTICE] mod_fsmux.c:95 event: channel_hangup 36e9ea37-5d94-439a-97be-c3d490db12d6 recv 519 bytes from udp/[AA.AA.AA.AA]:5060 at 13:18:29.543596: ------------------------------------------------------------------------ SIP/2.0 200 OK From: ;tag=U7c55U2Sva2pS To: ;tag=4dcbcc8-7f000001-13c4-15d31-2c2bf08a-15d31 Call-ID: 4e34100-7f000001-13c4-15d31-510bd4d5-15d31 at AA.AA.AA.AA CSeq: 113277690 BYE Via: SIP/2.0/UDP BB.BB.BB.BB:5062;rport=5062;branch=z9hG4bKUQmt9KNeg9yjN Supported: 100rel,replaces Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER User-Agent: ADTRAN_Total_Access_924e_2nd_Gen/R11.10.7.HA.E Content-Length: 0 ------------------------------------------------------------------------ 2017-10-05 13:18:30.508246 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:30.508246 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:30.508246 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. 2017-10-05 13:18:31.508234 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:31.508234 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:31.508234 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. 2017-10-05 13:18:32.508242 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:32.508242 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:32.508242 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. 2017-10-05 13:18:33.508234 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:33.508234 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:33.508234 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. 2017-10-05 13:18:34.508224 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:34.508224 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:34.508224 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. 2017-10-05 13:18:35.508246 [INFO] mod_fsmux.c:168 runtime is working 2017-10-05 13:18:35.508246 [INFO] mod_fsmux.c:173 show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_accountcode,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 36e9ea37-5d94-439a-97be-c3d490db12d6,inbound,2017-10-05 13:18:20,1507227500,sofia/internal/010001103 at AA.AA.AA.AA :5060,CS_EXECUTE,010001103,010001103,AA.AA.AA.AA,28,,,,ACTIVE,,,,, jbaik-dev.icsolutions.com,,,,,,,,,,,,,,,,,,,,,, 1 total. 2017-10-05 13:18:35.508246 [INFO] mod_fsmux.c:178 conference list +OK No active conferences. If I use API and conference hup then the call is cleared, but I want to know why switch_channel_hangup() call is not clearing the call. Thank you in advance! Jinserk -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: dummy.zip Type: application/zip Size: 250756 bytes Desc: not available URL: From panos at kioski.gr Fri Oct 6 05:12:27 2017 From: panos at kioski.gr (Panagiotis Voutskidis) Date: Fri, 6 Oct 2017 08:12:27 +0300 Subject: [Freeswitch-users] FreeSWITCH 1.8 In-Reply-To: References: Message-ID: <40eb3796-5428-2642-04fd-e4f322141b4e@kioski.gr> I can't find any details about version 1.8, so I will ask here: - When is the new version expected? - Are there any significant changes/new features? On 09/30/2017 03:24 AM, Oleg Stolyar wrote: > Thanks Jungle! > > On Fri, Sep 29, 2017 at 12:39 PM, jungle Boogie > > wrote: > > On 29 September 2017 at 06:44, Oleg Stolyar > wrote: > > Hi guys, > > > > sorry if I missed an announcement, but has version 1.8 been > officially > > released? > > > > It doesn't look like it. Look on the lower right, under the books and > you'll see 1.6.19 is the current version. > > https://freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 pskoul at gmail.com Fri Oct 6 06:11:28 2017 From: pskoul at gmail.com (Panagiotis Skoulikaritis) Date: Fri, 6 Oct 2017 09:11:28 +0300 Subject: [Freeswitch-users] Removal of transport=udp from the contact header in the FreeSWITCH replies Message-ID: <89bce6ea-2221-23f4-3e1a-a63c9444681c@gmail.com> Dear all Is it possible to remove the transport=udp from the contact header on the replies that the FreeSWITCH sends ? Best Regards Panagiotis Skoulikaritis From bilaln018 at gmail.com Fri Oct 6 06:59:51 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 6 Oct 2017 11:59:51 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: Anybody here that can please help me out? Regards Abbasi On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi wrote: > Hi Users, > I am using lua script to actually call the curl_sendfile, i am able to > successfully POST the file on URL, but i am only curious to know the > response variables to get the status. > Like i used curl previously and there are two variables that are auto set > curl_response_data and curl_response_code, i am looking same in the > curl_sendfile. > CAn anybody help me in this? > > P.S: i did tried to send file using curl, but i am not aware that how to > do that using the curl -F(--form) option, i can upload a file using > commandline linux curl command , but could not mapp the option -F in the > freeswitch. > > Regards > Abbasi > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ovoshlook at gmail.com Fri Oct 6 07:55:37 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Fri, 6 Oct 2017 10:55:37 +0300 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: First of all - be carefull with file uploading from the dialplan, because it can block call handling while you did not get response. Secondly - you also can use some external lua httpclient binding via require and use it to get response statuses that you need. On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: > Anybody here that can please help me out? > > Regards > Abbasi > > On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi wrote: > >> Hi Users, >> I am using lua script to actually call the curl_sendfile, i am able to >> successfully POST the file on URL, but i am only curious to know the >> response variables to get the status. >> Like i used curl previously and there are two variables that are auto set >> curl_response_data and curl_response_code, i am looking same in the >> curl_sendfile. >> CAn anybody help me in this? >> >> P.S: i did tried to send file using curl, but i am not aware that how to >> do that using the curl -F(--form) option, i can upload a file using >> commandline linux curl command , but could not mapp the option -F in the >> freeswitch. >> >> Regards >> Abbasi >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Oct 6 08:22:37 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 6 Oct 2017 08:22:37 +0000 Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> Hi All, We've been using mod_callcenter and have experienced issues with agents still receiving mod_callcenter calls when on outbound or internal calls handled by mod_sofia. The issue can be worked around by limiting the number of concurrent calls per extension and\or disabling call waiting but this doesn't solve the problem. Is there anyway we can skip busy agents using mod_callcenter when they're not on a mod_callcenter call? Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Oct 6 12:15:41 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 6 Oct 2017 12:15:41 +0000 Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A441F@mbx-01.sysconfig.co.uk> If this isn't possible which at this stage I presume is the case, perhaps we could use limit to keep count of the number of calls per extension. Then we just need a way for mod_callcenter to check the limit, I've tried using this in the agent contact string but spaces don't appear to be allowed so the contact string ends at hash. ${cond(${limit_usage(hash mydomain ext)} >= 1 ? error/user_busy : ${sofia_contact(*/ext at mydomain)})} Any ideas? Thanks, Shaun ________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk] Sent: 06 October 2017 09:22 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? Hi All, We've been using mod_callcenter and have experienced issues with agents still receiving mod_callcenter calls when on outbound or internal calls handled by mod_sofia. The issue can be worked around by limiting the number of concurrent calls per extension and\or disabling call waiting but this doesn't solve the problem. Is there anyway we can skip busy agents using mod_callcenter when they're not on a mod_callcenter call? Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Fri Oct 6 13:06:49 2017 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Fri, 6 Oct 2017 15:06:49 +0200 Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A441F@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A441F@mbx-01.sysconfig.co.uk> Message-ID: <9b4f1618-c591-9dad-dfc7-27f8ab2e593c@wirelessmundi.com> Possible solution: more details in: https://freeswitch.org/jira/browse/FS-9609 On 10/06/2017 02:15 PM, Shaun Stokes wrote: > If this isn't possible which at this stage I presume is the case, > perhaps we could use limit to keep count of the number of calls per > extension. > > Then we just need a way for mod_callcenter to check the limit, I've > tried using this in the agent contact string but spaces don't appear > to be allowed so the contact string ends at hash. > ${cond(${limit_usage(hash mydomain ext)} >= 1 ? error/user_busy : > ${sofia_contact(*/ext at mydomain)})} > > Any ideas? > > Thanks, > Shaun > ------------------------------------------------------------------------ > *From:* FreeSWITCH-users > [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Shaun > Stokes [shaun.stokes at itec-support.co.uk] > *Sent:* 06 October 2017 09:22 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] mod_callcenter - How to skip busy agents > not on a mod_callcenter call? > > Hi All, > > We've been using mod_callcenter and have experienced issues with > agents still receiving mod_callcenter calls when on outbound or > internal calls handled by mod_sofia. > > The issue can be worked around by limiting the number of concurrent > calls per extension and\or disabling call waiting but this doesn't > solve the problem. > > Is there anyway we can skip busy agents using mod_callcenter when > they're not on a mod_callcenter call? > > Thanks, > Shaun > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs > Virus Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From freeswitch940 at gmail.com Fri Oct 6 13:30:24 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Fri, 06 Oct 2017 13:30:24 +0000 Subject: [Freeswitch-users] Video calls In-Reply-To: References: <0ff56944-6a38-5c16-5a41-ed189a6fc40a@wirelessmundi.com> Message-ID: Anybody can help? On Wed, 4 Oct 2017 at 6:08 PM, Freeswitch user wrote: > Thanks For Reply. I'm trying the same which is mentioned in provided url > but getting some error.. > > DialPlan : > > > > > > > > > > Error:- 2017-10-04 09:33:22.678400 [ERR] mod_fsv.c:969 You are asking to > write 2048 bytes of data which is not supported. Please set > enable_file_write_buffering=false to use .fsv format > 2017-10-04 09:33:22.678400 [ERR] switch_ivr_async.c:1169 Error writing > /tmp/firstTest.fsv > 2017-10-04 09:33:22.938385 [ERR] mod_fsv.c:969 You are asking to write > 2048 bytes of data which is not supported. Please set > enable_file_write_buffering=false to use .fsv format > 2017-10-04 09:33:22.938385 [ERR] switch_ivr_async.c:1169 Error writing > /tmp/firstTest.fsv > 2017-10-04 09:33:23.198409 [ERR] mod_fsv.c:969 You are asking to write > 2048 bytes of data which is not supported. Please set > enable_file_write_buffering=false to use .fsv format > > > > On Wed, Oct 4, 2017 at 4:42 PM, António Silva > wrote: > >> yes you can, check: >> https://freeswitch.org/confluence/display/FREESWITCH/Video-recording >> >> On 10/04/2017 01:06 PM, Freeswitch user wrote: >> >> Dear users, >> >> Can we record the video calls in freeswitch. Extension A and extension B >> having a video call by freeswitch. So can we record it? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/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 bilaln018 at gmail.com Fri Oct 6 14:04:42 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 6 Oct 2017 19:04:42 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: Hi, Thanks for the answer, 1) Yes i am aware of that, in fact we need to inform the user that file is uploaded successfully or not, so it's normal to have few seconds delay. 2)Yes i tried that way, and i did able to do that using lua cURL library, but i am only interested to know that may be the native function can provide me this, I mean if somebody has build this module there should/must be a way to get the status. Regards Abbasi On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko wrote: > First of all - be carefull with file uploading from the dialplan, because > it can block call handling while you did not get response. > > Secondly - you also can use some external lua httpclient binding via > require and use it to get response statuses that you need. > > On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: > >> Anybody here that can please help me out? >> >> Regards >> Abbasi >> >> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi wrote: >> >>> Hi Users, >>> I am using lua script to actually call the curl_sendfile, i am able to >>> successfully POST the file on URL, but i am only curious to know the >>> response variables to get the status. >>> Like i used curl previously and there are two variables that are auto >>> set curl_response_data and curl_response_code, i am looking same in the >>> curl_sendfile. >>> CAn anybody help me in this? >>> >>> P.S: i did tried to send file using curl, but i am not aware that how to >>> do that using the curl -F(--form) option, i can upload a file using >>> commandline linux curl command , but could not mapp the option -F in the >>> freeswitch. >>> >>> Regards >>> Abbasi >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 Fri Oct 6 14:03:52 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 6 Oct 2017 14:03:52 +0000 Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? In-Reply-To: <9b4f1618-c591-9dad-dfc7-27f8ab2e593c@wirelessmundi.com> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A441F@mbx-01.sysconfig.co.uk>, <9b4f1618-c591-9dad-dfc7-27f8ab2e593c@wirelessmundi.com> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A466A@mbx-01.sysconfig.co.uk> This would be a great solution, however just tried testing and it doesn't currently work in our environment on FreeSWITCH 1.6.19 we get "Invalid Application callcenter_track". I can see the changes have been committed: https://freeswitch.org/fisheye/changelog/freeswitch?cs=15a232b5bb4e2a0f83c84469b549c9adcc1ce280 We'll do a bit more digging and will post back with the results. Thanks, Shaun ________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of António Silva [asilva at wirelessmundi.com] Sent: 06 October 2017 14:06 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? Possible solution: more details in: https://freeswitch.org/jira/browse/FS-9609 On 10/06/2017 02:15 PM, Shaun Stokes wrote: If this isn't possible which at this stage I presume is the case, perhaps we could use limit to keep count of the number of calls per extension. Then we just need a way for mod_callcenter to check the limit, I've tried using this in the agent contact string but spaces don't appear to be allowed so the contact string ends at hash. ${cond(${limit_usage(hash mydomain ext)} >= 1 ? error/user_busy : ${sofia_contact(*/ext at mydomain)})} Any ideas? Thanks, Shaun ________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk] Sent: 06 October 2017 09:22 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? Hi All, We've been using mod_callcenter and have experienced issues with agents still receiving mod_callcenter calls when on outbound or internal calls handled by mod_sofia. The issue can be worked around by limiting the number of concurrent calls per extension and\or disabling call waiting but this doesn't solve the problem. Is there anyway we can skip busy agents using mod_callcenter when they're not on a mod_callcenter call? Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Fri Oct 6 14:09:26 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 6 Oct 2017 18:09:26 +0400 Subject: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on windows Message-ID: hi, i have two instances of FS running on a windows server, one is used for routing to carriers and that uses xml files and that works fine, the second instance is set up to allow clients to register to it using tls and srtp and users are authenticated using xml_curl which calls a php script which inturn sends the directory users details on a register from a client. The problem is every few hours or so FS stops accepting new clients unless i restart the service, i check the php script and the webserver and those are running just fine so no idea whats causing FS to stop calling the script or something else. -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Fri Oct 6 13:39:21 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Fri, 6 Oct 2017 13:39:21 +0000 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Have you checked if Nginx has timeout settings and/or keep alive settings? https://stackoverflow.com/questions/10550558/nginx-tcp-websockets-timeout-keepalive-config Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Thursday, October 5, 2017 11:07 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Sending keepalives on Verto websocket I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality? ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ovoshlook at gmail.com Fri Oct 6 14:20:53 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Fri, 6 Oct 2017 17:20:53 +0300 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: i think in your case will be better to use bg_api command and run handler there. Regarding native or not. Here is a good question - what will be faster and i not sure that native will be faster way than lua binding On Oct 6, 2017 17:05, "Bilal Abbasi" wrote: > Hi, > Thanks for the answer, > 1) Yes i am aware of that, in fact we need to inform the user that file is > uploaded successfully or not, so it's normal to have few seconds delay. > 2)Yes i tried that way, and i did able to do that using lua cURL library, > but i am only interested to know that may be the native function can > provide me this, I mean if somebody has build this module there should/must > be a way to get the status. > > Regards > Abbasi > > On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko > wrote: > >> First of all - be carefull with file uploading from the dialplan, because >> it can block call handling while you did not get response. >> >> Secondly - you also can use some external lua httpclient binding via >> require and use it to get response statuses that you need. >> >> On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: >> >>> Anybody here that can please help me out? >>> >>> Regards >>> Abbasi >>> >>> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi >>> wrote: >>> >>>> Hi Users, >>>> I am using lua script to actually call the curl_sendfile, i am able to >>>> successfully POST the file on URL, but i am only curious to know the >>>> response variables to get the status. >>>> Like i used curl previously and there are two variables that are auto >>>> set curl_response_data and curl_response_code, i am looking same in the >>>> curl_sendfile. >>>> CAn anybody help me in this? >>>> >>>> P.S: i did tried to send file using curl, but i am not aware that how >>>> to do that using the curl -F(--form) option, i can upload a file using >>>> commandline linux curl command , but could not mapp the option -F in the >>>> freeswitch. >>>> >>>> Regards >>>> Abbasi >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 bilaln018 at gmail.com Fri Oct 6 15:01:37 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 6 Oct 2017 20:01:37 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: Thanks i will try that way. Highly appreciated your response. Regards Abbasi On Fri, Oct 6, 2017 at 7:20 PM, Yuriy Gorlichenko wrote: > i think in your case will be better to use bg_api command and run handler > there. > Regarding native or not. Here is a good question - what will be faster and > i not sure that native will be faster way than lua binding > > On Oct 6, 2017 17:05, "Bilal Abbasi" wrote: > >> Hi, >> Thanks for the answer, >> 1) Yes i am aware of that, in fact we need to inform the user that file >> is uploaded successfully or not, so it's normal to have few seconds delay. >> 2)Yes i tried that way, and i did able to do that using lua cURL library, >> but i am only interested to know that may be the native function can >> provide me this, I mean if somebody has build this module there should/must >> be a way to get the status. >> >> Regards >> Abbasi >> >> On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko >> wrote: >> >>> First of all - be carefull with file uploading from the dialplan, >>> because it can block call handling while you did not get response. >>> >>> Secondly - you also can use some external lua httpclient binding via >>> require and use it to get response statuses that you need. >>> >>> On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: >>> >>>> Anybody here that can please help me out? >>>> >>>> Regards >>>> Abbasi >>>> >>>> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi >>>> wrote: >>>> >>>>> Hi Users, >>>>> I am using lua script to actually call the curl_sendfile, i am able to >>>>> successfully POST the file on URL, but i am only curious to know the >>>>> response variables to get the status. >>>>> Like i used curl previously and there are two variables that are auto >>>>> set curl_response_data and curl_response_code, i am looking same in the >>>>> curl_sendfile. >>>>> CAn anybody help me in this? >>>>> >>>>> P.S: i did tried to send file using curl, but i am not aware that how >>>>> to do that using the curl -F(--form) option, i can upload a file using >>>>> commandline linux curl command , but could not mapp the option -F in the >>>>> freeswitch. >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 gregor at infomedia.si Fri Oct 6 16:14:08 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 6 Oct 2017 18:14:08 +0200 Subject: [Freeswitch-users] mod_callcenter - How to skip busy agents not on a mod_callcenter call? In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A466A@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A412C@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A441F@mbx-01.sysconfig.co.uk> <9b4f1618-c591-9dad-dfc7-27f8ab2e593c@wirelessmundi.com> <6FD2F8B5BB72834E9939AEDF9FB802A901E86A466A@mbx-01.sysconfig.co.uk> Message-ID: This is implemented in 1.9 2017-10-06 16:03 GMT+02:00 Shaun Stokes : > This would be a great solution, however just tried testing and it doesn't > currently work in our environment on FreeSWITCH 1.6.19 we get "Invalid > Application callcenter_track". > > I can see the changes have been committed: https://freeswitch. > org/fisheye/changelog/freeswitch?cs=15a232b5bb4e2a0f83c84469b549c9 > adcc1ce280 > > We'll do a bit more digging and will post back with the results. > > Thanks, > Shaun > > ------------------------------ > *From:* FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] > on behalf of António Silva [asilva at wirelessmundi.com] > *Sent:* 06 October 2017 14:06 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] mod_callcenter - How to skip busy > agents not on a mod_callcenter call? > > Possible solution: > > > > > more details in: > > https://freeswitch.org/jira/browse/FS-9609 > > > > On 10/06/2017 02:15 PM, Shaun Stokes wrote: > > If this isn't possible which at this stage I presume is the case, perhaps > we could use limit to keep count of the number of calls per extension. > > Then we just need a way for mod_callcenter to check the limit, I've tried > using this in the agent contact string but spaces don't appear to be > allowed so the contact string ends at hash. > ${cond(${limit_usage(hash mydomain ext)} >= 1 ? error/user_busy : > ${sofia_contact(*/ext at mydomain)})} > > Any ideas? > > Thanks, > Shaun > ------------------------------ > *From:* FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] > on behalf of Shaun Stokes [shaun.stokes at itec-support.co.uk] > *Sent:* 06 October 2017 09:22 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] mod_callcenter - How to skip busy agents > not on a mod_callcenter call? > > Hi All, > > We've been using mod_callcenter and have experienced issues with agents > still receiving mod_callcenter calls when on outbound or internal calls > handled by mod_sofia. > > The issue can be worked around by limiting the number of concurrent calls > per extension and\or disabling call waiting but this doesn't solve the > problem. > > Is there anyway we can skip busy agents using mod_callcenter when they're > not on a mod_callcenter call? > > Thanks, > Shaun > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > Saludos / Regards / Cumprimentos > António Silva > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Fri Oct 6 17:50:43 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Fri, 6 Oct 2017 17:50:43 +0000 Subject: [Freeswitch-users] session id (rfc7329) support (VoIP and Verto)? Message-ID: Does FreeSWITCH have session id (rfc7329) support (VoIP and Verto)? Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From siju.irs at gmail.com Fri Oct 6 15:49:33 2017 From: siju.irs at gmail.com (Siju Nair) Date: Fri, 6 Oct 2017 21:19:33 +0530 Subject: [Freeswitch-users] Fusion pbx Message-ID: <6BD940D7-A6B3-4BFA-8EAE-755CB9149FBC@gmail.com> Hi team I need to install fusion pbx, since am new to this I don’t whether I need to install freeswitch separately or while installing fusion pbx, FS also will be installed... plz help !! Sent from my iPhone From petrpwp at gmail.com Fri Oct 6 18:50:26 2017 From: petrpwp at gmail.com (Pete Procenko) Date: Fri, 6 Oct 2017 21:50:26 +0300 Subject: [Freeswitch-users] Fusion pbx In-Reply-To: <6BD940D7-A6B3-4BFA-8EAE-755CB9149FBC@gmail.com> References: <6BD940D7-A6B3-4BFA-8EAE-755CB9149FBC@gmail.com> Message-ID: Just follow the instructions on fusionpbx.com, they will install freeswitch from official repo themselves, if You will use their automated install script There is no support for fusionpbx here. 2017-10-06 18:49 GMT+03:00 Siju Nair : > Hi team > I need to install fusion pbx, since am new to this I don’t whether I need > to install freeswitch separately or while installing fusion pbx, FS also > will be installed... plz help !! > > Sent from my iPhone > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 chad at apartmentlines.com Fri Oct 6 20:52:52 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Fri, 6 Oct 2017 13:52:52 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: There are various suggested solutions out there. On the Nginx config side, easiest is to increase the proxy_read_timeout setting to something high, but that seems like it could result in connections getting hung open if the server process goes away. Then there’s this module which looks pretty fancy and claims to solve the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module — it requires custom compiling nginx though… I think I’ll stick with my current solution, which basically follows the other suggestion of having the server send periodic keepalives, except that I’m initiating from the client. Steps for others interested: 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant domains in your user directory 2. Do something like this after a successful websocket connection: MODULE.keepAliveTimer = setInterval(function() { verto.rpcClient.call("echo", {keepalive: true}); }, VERTO_KEEPALIVE_INTERVAL) 3. Add some cleanup on websocket close: if (MODULE.keepAliveTimer) { clearInterval(MODULE.keepAliveTimer); MODULE.keepAliveTimer = null; } Request and response lengths are both under 150 characters, so pretty short, and I set the intervals at 50 seconds, since Nginx’s default timeout is 60 seconds. On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert wrote: > Have you checked if Nginx has timeout settings and/or keep alive settings? > > > > https://stackoverflow.com/questions/10550558/nginx-tcp- > websockets-timeout-keepalive-config > > > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Thursday, October 5, 2017 11:07 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket > > > > I've switched to using Nginx to proxy Verto websockets, and have run into > a small snag: by default, if Nginx doesn't read any data from a proxy > backend within 60 seconds, it closes the connection, even for websockets. > > > > It appears the recommended solution is to have the server send some kind > of regular keepalive. I poked around in mod_verto.c and found a > 'request.keepalive' variable, but I'm unclear how to set that in the > request, and/or if it even accomplishes what I'm wanting. > > > > I've solved the issue for now by periodically sending a JSON RPC 'echo' > request along the websocket every 50 seconds, and the reply from the server > is enough to keep the connection open. This is fine, but I am curious if > there's a way to do it just from the server side, and if not, if it's worth > it to add a setting to enable that functionality? > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 bilaln018 at gmail.com Sat Oct 7 18:05:45 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 7 Oct 2017 23:05:45 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: Hi, I build the lua script and used the cURL module for that( https://github.com/Lua-cURL/Lua-cURLv3) I am successfully place the POST request using that module, but when i use that with in freeswitch using application lua it raise the ERR 2017-10-07 17:46:47.592484 [ERR] mod_lua.cpp:203 error loading module 'lcurl' from file '/usr/local/lib/lua/5.1/lcurl.so': /usr/local/lib/lua/5.1/lcurl.so: undefined symbol: lua_tointeger I am not sure why its raising this error while loading, its not raising this when i run the script directly fromt the shell. Can you please help me in this regard. Regards Abbasi On Fri, Oct 6, 2017 at 8:01 PM, Bilal Abbasi wrote: > Thanks i will try that way. Highly appreciated your response. > > Regards > Abbasi > > On Fri, Oct 6, 2017 at 7:20 PM, Yuriy Gorlichenko > wrote: > >> i think in your case will be better to use bg_api command and run handler >> there. >> Regarding native or not. Here is a good question - what will be faster >> and i not sure that native will be faster way than lua binding >> >> On Oct 6, 2017 17:05, "Bilal Abbasi" wrote: >> >>> Hi, >>> Thanks for the answer, >>> 1) Yes i am aware of that, in fact we need to inform the user that file >>> is uploaded successfully or not, so it's normal to have few seconds delay. >>> 2)Yes i tried that way, and i did able to do that using lua cURL >>> library, but i am only interested to know that may be the native function >>> can provide me this, I mean if somebody has build this module there >>> should/must be a way to get the status. >>> >>> Regards >>> Abbasi >>> >>> On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko >>> wrote: >>> >>>> First of all - be carefull with file uploading from the dialplan, >>>> because it can block call handling while you did not get response. >>>> >>>> Secondly - you also can use some external lua httpclient binding via >>>> require and use it to get response statuses that you need. >>>> >>>> On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: >>>> >>>>> Anybody here that can please help me out? >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>>> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi >>>>> wrote: >>>>> >>>>>> Hi Users, >>>>>> I am using lua script to actually call the curl_sendfile, i am able >>>>>> to successfully POST the file on URL, but i am only curious to know the >>>>>> response variables to get the status. >>>>>> Like i used curl previously and there are two variables that are auto >>>>>> set curl_response_data and curl_response_code, i am looking same in the >>>>>> curl_sendfile. >>>>>> CAn anybody help me in this? >>>>>> >>>>>> P.S: i did tried to send file using curl, but i am not aware that how >>>>>> to do that using the curl -F(--form) option, i can upload a file using >>>>>> commandline linux curl command , but could not mapp the option -F in the >>>>>> freeswitch. >>>>>> >>>>>> Regards >>>>>> Abbasi >>>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Sat Oct 7 04:40:40 2017 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Fri, 6 Oct 2017 23:40:40 -0500 Subject: [Freeswitch-users] Delay Bye Message Message-ID: Hi. I have what appears to be an unusual request. One of my clients has a clause in his contract with the local telco/sip provider that requires the majority of calls to be at or greater than 12 seconds. On a former platform, he has an option called 'Delayed Bye Message.' This option, if checked, allows him to specify how many seconds to delay hanging up the call once leg B initiates the request. So, all calls will last at least 12 seconds. Does anyone know how to do that with Freeswitch? Any help is greatly appreciated. Thank you, Alex Perkins -------------- next part -------------- An HTML attachment was scrubbed... URL: From arsen.semionov at gmail.com Fri Oct 6 18:48:45 2017 From: arsen.semionov at gmail.com (Arsen) Date: Fri, 6 Oct 2017 21:48:45 +0300 Subject: [Freeswitch-users] Fusion pbx In-Reply-To: <6BD940D7-A6B3-4BFA-8EAE-755CB9149FBC@gmail.com> References: <6BD940D7-A6B3-4BFA-8EAE-755CB9149FBC@gmail.com> Message-ID: Hi, FS will be installed. Here is the script. https://www.fusionpbx.com/download.php Regards, Arsen On Fri, Oct 6, 2017 at 6:49 PM, Siju Nair wrote: > Hi team > I need to install fusion pbx, since am new to this I don’t whether I need > to install freeswitch separately or while installing fusion pbx, FS also > will be installed... plz help !! > > Sent from my iPhone > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 dayton at voxter.ca Fri Oct 6 19:53:16 2017 From: dayton at voxter.ca (Dayton Turner) Date: Fri, 06 Oct 2017 12:53:16 -0700 Subject: [Freeswitch-users] Choppy and fragmented return-audio problem using Unicast Message-ID: <59D7DF2C.3010904@voxter.com> Hello list, I've got a question about using the unicast call command via ESL. I realize there are other ways to do this, such as writing a native C module, but with a lack of developers with the necessary C skills, we're trying to take a "simpler to implement" approach at first just to get things rolling. The objective is to send and receive audio to a 3rd party application. In this case, we're using a UDP transport via unicast, running a small application written in golang on the FS server itself. As a proof of concept, we're simply capturing received RTP on the UDP socket, and piping it back to the UDP port FS is listening on - basically an echo test. Audio does come in, we are able to send audio back, but the return audio as we hear it sounds incredibly choppy, and like some of the audio frames are being clipped or dropped entirely. We also tested merely saving the received audio to a file and wrote it to disk. Loading it up in audacity and importing raw audio, it plays back properly when we import it as ULAW 8khz mono. Additionally, we also tried skipping the echo test and instead simply opening the saved file and bitstreaming it back to FS on the listening unicast UDP port. Despite playing back properly in audacity, its all choppy and cut up when played back via the UDP stream. Nothing fancy going on, literally just opening the file and bitstreaming it to the UDP socket. We also tried a couple example 8khz ulaw files, just streaming those back and they behave similarly. In fact, they sound sped up (in playback "speed", but not pitch, which is interesting). Am I missing something about what format the UDP unicast listening port in FS expects to receive, differently than what it sent out in the first place? As we receive 8khz ulaw, we're sending that back in return. flags:native is specified on the ESL call-command. We also tried transcoding to L16 before sending it back, no luck there either (completely unintelligible stream of fuzz) I thought about trying another method of allowing a 3rd party application to access the audio stream (both receive and send back) like mod_tts_commandline but it wasnt clear wether this route permits for audio in both directions to be involved or if it was merely return audio. Additionally, via ESL if we execute the playback application and just ask FS to play one of its own soundfiles, it sounds perfectly fine, so Im confident that the setup up until the unicast test is working correctly. Testing is on FS 1.6.15 on Centos 7. Any thoughts? Thanks in advance :) Dayton -------------- next part -------------- An HTML attachment was scrubbed... URL: From ovoshlook at gmail.com Sat Oct 7 19:25:05 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Sat, 7 Oct 2017 22:25:05 +0300 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: You installed lib for lua 5.1 but Freeswitch requires lua 5.2 you soulh it to install via luarocks for 5.2 2017-10-07 21:05 GMT+03:00 Bilal Abbasi : > Hi, > I build the lua script and used the cURL module for that( > https://github.com/Lua-cURL/Lua-cURLv3) > I am successfully place the POST request using that module, but when i use > that with in freeswitch using application lua it raise the ERR > > 2017-10-07 17:46:47.592484 [ERR] mod_lua.cpp:203 error loading module > 'lcurl' from file '/usr/local/lib/lua/5.1/lcurl.so': > > /usr/local/lib/lua/5.1/lcurl.so: undefined symbol: lua_tointeger > > > I am not sure why its raising this error while loading, its not raising > this when i run the script directly fromt the shell. > > Can you please help me in this regard. > > > Regards > > Abbasi > > On Fri, Oct 6, 2017 at 8:01 PM, Bilal Abbasi wrote: > >> Thanks i will try that way. Highly appreciated your response. >> >> Regards >> Abbasi >> >> On Fri, Oct 6, 2017 at 7:20 PM, Yuriy Gorlichenko >> wrote: >> >>> i think in your case will be better to use bg_api command and run >>> handler there. >>> Regarding native or not. Here is a good question - what will be faster >>> and i not sure that native will be faster way than lua binding >>> >>> On Oct 6, 2017 17:05, "Bilal Abbasi" wrote: >>> >>>> Hi, >>>> Thanks for the answer, >>>> 1) Yes i am aware of that, in fact we need to inform the user that file >>>> is uploaded successfully or not, so it's normal to have few seconds delay. >>>> 2)Yes i tried that way, and i did able to do that using lua cURL >>>> library, but i am only interested to know that may be the native function >>>> can provide me this, I mean if somebody has build this module there >>>> should/must be a way to get the status. >>>> >>>> Regards >>>> Abbasi >>>> >>>> On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko >>> > wrote: >>>> >>>>> First of all - be carefull with file uploading from the dialplan, >>>>> because it can block call handling while you did not get response. >>>>> >>>>> Secondly - you also can use some external lua httpclient binding via >>>>> require and use it to get response statuses that you need. >>>>> >>>>> On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: >>>>> >>>>>> Anybody here that can please help me out? >>>>>> >>>>>> Regards >>>>>> Abbasi >>>>>> >>>>>> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi >>>>>> wrote: >>>>>> >>>>>>> Hi Users, >>>>>>> I am using lua script to actually call the curl_sendfile, i am able >>>>>>> to successfully POST the file on URL, but i am only curious to know the >>>>>>> response variables to get the status. >>>>>>> Like i used curl previously and there are two variables that are >>>>>>> auto set curl_response_data and curl_response_code, i am looking same in >>>>>>> the curl_sendfile. >>>>>>> CAn anybody help me in this? >>>>>>> >>>>>>> P.S: i did tried to send file using curl, but i am not aware that >>>>>>> how to do that using the curl -F(--form) option, i can upload a file using >>>>>>> commandline linux curl command , but could not mapp the option -F in the >>>>>>> freeswitch. >>>>>>> >>>>>>> Regards >>>>>>> Abbasi >>>>>>> >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 bipin at xbipin.com Sat Oct 7 19:51:54 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sat, 07 Oct 2017 23:51:54 +0400 Subject: [Freeswitch-users] Delay Bye Message In-Reply-To: References: Message-ID: <15ef864df90.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, I don't think there is anything in FS that can do that, at least that's what I was told a few years back. On October 7, 2017 11:05:32 PM Alexander Perkins wrote: > Hi. I have what appears to be an unusual request. > > One of my clients has a clause in his contract with the local telco/sip > provider that requires the majority of calls to be at or greater than 12 > seconds. On a former platform, he has an option called 'Delayed Bye > Message.' This option, if checked, allows him to specify how many seconds > to delay hanging up the call once leg B initiates the request. So, all > calls will last at least 12 seconds. Does anyone know how to do that with > Freeswitch? Any help is greatly appreciated. > > Thank you, > Alex Perkins > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 krice at freeswitch.org Sat Oct 7 19:56:55 2017 From: krice at freeswitch.org (Ken Rice) Date: Sat, 7 Oct 2017 14:56:55 -0500 Subject: [Freeswitch-users] Delay Bye Message In-Reply-To: References: Message-ID: no you cant just delay the bye message to avoid short call duration penalties. it doesnt work that way. Sent from my iPhone > On Oct 6, 2017, at 23:40, Alexander Perkins wrote: > > Hi. I have what appears to be an unusual request. > > One of my clients has a clause in his contract with the local telco/sip provider that requires the majority of calls to be at or greater than 12 seconds. On a former platform, he has an option called 'Delayed Bye Message.' This option, if checked, allows him to specify how many seconds to delay hanging up the call once leg B initiates the request. So, all calls will last at least 12 seconds. Does anyone know how to do that with Freeswitch? Any help is greatly appreciated. > > Thank you, > Alex Perkins > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rbetancor at gmail.com Sat Oct 7 20:01:32 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Sat, 7 Oct 2017 21:01:32 +0100 Subject: [Freeswitch-users] Delay Bye Message In-Reply-To: References: Message-ID: If the B leg hangs you, it doesn't matter how long you try to delay the ACK ... from provider point of view, it got hanged when they saw the BYE from B leg. If your FS is the one that is going to hang, obviously, you could delay your BYE 2017-10-07 20:56 GMT+01:00 Ken Rice : > no you cant just delay the bye message to avoid short call duration > penalties. it doesnt work that way. > > Sent from my iPhone > > On Oct 6, 2017, at 23:40, Alexander Perkins com> wrote: > > Hi. I have what appears to be an unusual request. > > One of my clients has a clause in his contract with the local telco/sip > provider that requires the majority of calls to be at or greater than 12 > seconds. On a former platform, he has an option called 'Delayed Bye > Message.' This option, if checked, allows him to specify how many seconds > to delay hanging up the call once leg B initiates the request. So, all > calls will last at least 12 seconds. Does anyone know how to do that with > Freeswitch? Any help is greatly appreciated. > > Thank you, > Alex Perkins > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 ssinyagin at gmail.com Sat Oct 7 21:05:05 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 7 Oct 2017 23:05:05 +0200 Subject: [Freeswitch-users] Delay Bye Message In-Reply-To: References: Message-ID: If you bridge a call, and your user hangs up, you you can send silence to the other leg and hang-up after a timer. This would increase the call duration, but cause inconvenience to your user's counterparty. On 7 Oct 2017 21:04, "Alexander Perkins" wrote: > Hi. I have what appears to be an unusual request. > > One of my clients has a clause in his contract with the local telco/sip > provider that requires the majority of calls to be at or greater than 12 > seconds. On a former platform, he has an option called 'Delayed Bye > Message.' This option, if checked, allows him to specify how many seconds > to delay hanging up the call once leg B initiates the request. So, all > calls will last at least 12 seconds. Does anyone know how to do that with > Freeswitch? Any help is greatly appreciated. > > Thank you, > Alex Perkins > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rick at magicmail.mooo.com Sun Oct 8 14:50:45 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Sun, 8 Oct 2017 15:50:45 +0100 Subject: [Freeswitch-users] Issue with database updates Message-ID: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols ) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!? -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Sun Oct 8 15:41:10 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sun, 8 Oct 2017 17:41:10 +0200 Subject: [Freeswitch-users] Choppy and fragmented return-audio problem using Unicast In-Reply-To: <59D7DF2C.3010904@voxter.com> References: <59D7DF2C.3010904@voxter.com> Message-ID: <2D82A1C0-6836-4155-89FF-A22341CF325F@vallimamod.org> Hi, I think you are getting a codec mismatch. Here is how the unicast command works according to the source code: - If native flag is not set, fs will use the L16 codec to exchange media beetween the session and the unicast socket. You can search for "Raw Codec Activation Success L16@" in the debug log to confirm it. - If the native flag is set, fs will use the same read and write codec as your current session: the rtp buffer is sent as-is to the unicast socket and the data received from the unicast socket is also sent back as-is to the session. Hope this helps! Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 6 Oct 2017, at 21:53, Dayton Turner wrote: > > Hello list, > > I've got a question about using the unicast call command via ESL. I realize there are other ways to do this, such as writing a native C module, but with a lack of developers with the necessary C skills, we're trying to take a "simpler to implement" approach at first just to get things rolling. > > The objective is to send and receive audio to a 3rd party application. In this case, we're using a UDP transport via unicast, running a small application written in golang on the FS server itself. As a proof of concept, we're simply capturing received RTP on the UDP socket, and piping it back to the UDP port FS is listening on - basically an echo test. > > Audio does come in, we are able to send audio back, but the return audio as we hear it sounds incredibly choppy, and like some of the audio frames are being clipped or dropped entirely. We also tested merely saving the received audio to a file and wrote it to disk. Loading it up in audacity and importing raw audio, it plays back properly when we import it as ULAW 8khz mono. Additionally, we also tried skipping the echo test and instead simply opening the saved file and bitstreaming it back to FS on the listening unicast UDP port. Despite playing back properly in audacity, its all choppy and cut up when played back via the UDP stream. Nothing fancy going on, literally just opening the file and bitstreaming it to the UDP socket. > > We also tried a couple example 8khz ulaw files, just streaming those back and they behave similarly. In fact, they sound sped up (in playback "speed", but not pitch, which is interesting). > > Am I missing something about what format the UDP unicast listening port in FS expects to receive, differently than what it sent out in the first place? As we receive 8khz ulaw, we're sending that back in return. flags:native is specified on the ESL call-command. We also tried transcoding to L16 before sending it back, no luck there either (completely unintelligible stream of fuzz) > > I thought about trying another method of allowing a 3rd party application to access the audio stream (both receive and send back) like mod_tts_commandline but it wasnt clear wether this route permits for audio in both directions to be involved or if it was merely return audio. > > Additionally, via ESL if we execute the playback application and just ask FS to play one of its own soundfiles, it sounds perfectly fine, so Im confident that the setup up until the unicast test is working correctly. > > Testing is on FS 1.6.15 on Centos 7. > > Any thoughts? Thanks in advance :) > > Dayton > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vma at vallimamod.org Sun Oct 8 18:05:51 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sun, 8 Oct 2017 20:05:51 +0200 Subject: [Freeswitch-users] Issue with database updates In-Reply-To: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> References: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> Message-ID: <959A217B-65B9-4413-B12E-C6A880F68334@vallimamod.org> Hi, I am not sure how you are updating these columns but if you are doing it directly through sql, I don't think it's the correct way. To set or update a value on a "presence data" column, you have to define or update a channel variable with the same name as the column. If there is no variable defined, the corresponding column will be set to NULL on the state change. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr tel: +33 6 62 60 68 97 . > On 8 Oct 2017, at 16:50, Rick Jarvis wrote: > > Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols ) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!? -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Sun Oct 8 18:13:26 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Sun, 8 Oct 2017 19:13:26 +0100 Subject: [Freeswitch-users] Issue with database updates In-Reply-To: <959A217B-65B9-4413-B12E-C6A880F68334@vallimamod.org> References: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> <959A217B-65B9-4413-B12E-C6A880F68334@vallimamod.org> Message-ID: <7197041B-AD3C-4018-AB27-8B7B77D3B45D@magicmail.mooo.com> I had resorted to updating the table directly through SQL as FS doesn't appear to dynamically update the table if I use uuid_setvar :/ > On 8 Oct 2017, at 19:05, Vallimamod Abdullah wrote: > > Hi, > > I am not sure how you are updating these columns but if you are doing it directly through sql, I don't think it's the correct way. > To set or update a value on a "presence data" column, you have to define or update a channel variable with the same name as the column. If there is no variable defined, the corresponding column will be set to NULL on the state change. > > Hope this helps. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > tel: +33 6 62 60 68 97 > . > >> On 8 Oct 2017, at 16:50, Rick Jarvis > wrote: >> >> Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols ) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 josedavid at zennio.com Mon Oct 9 05:51:30 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Mon, 9 Oct 2017 07:51:30 +0200 Subject: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on windows In-Reply-To: References: Message-ID: Hi Bipin, I've implemented a similar solution and it works correctly (without tls). If you want, give me more information (configuration, script, etc) and try to see if I find something strange. 2017-10-06 16:09 GMT+02:00 Bipin Patel : > hi, > > i have two instances of FS running on a windows server, one is used for > routing to carriers and that uses xml files and that works fine, the second > instance is set up to allow clients to register to it using tls and srtp > and users are authenticated using xml_curl which calls a php script which > inturn sends the directory users details on a register from a client. The > problem is every few hours or so FS stops accepting new clients unless i > restart the service, i check the php script and the webserver and those are > running just fine so no idea whats causing FS to stop calling the script or > something else. > > > -- > Regards, > Bipin > > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image006.jpg Type: image/jpeg Size: 1135 bytes Desc: not available URL: From raman.chv at gmail.com Mon Oct 9 06:44:01 2017 From: raman.chv at gmail.com (Ram) Date: Mon, 9 Oct 2017 12:14:01 +0530 Subject: [Freeswitch-users] Video call issue from webrtc to sip In-Reply-To: References: Message-ID: Hi, Any guidance to proceed further on this issue Regards Raman On Tue, Oct 3, 2017 at 7:59 PM, Ram wrote: > > Hi, > > Configured freeswitch as a media gateway between sip and web clients. > > When video call initiated from Sip client to web-rtc clients, audio sdp is > dropped when responding 200 ok to sip client after getting 200 ok response > from webrtc clients. where as video sdp is fine in 200 ok. > > From the logs found "marking rejected media" might causing issue. but not > getting clue for what is causing issue for rejected media. Enclosed > complete log for reference. > > soa_static.c:1189 offer_answer_step() soa_static(0x7ff8b0038c40, > soa_generate_answer): generating local description > soa_static.c:1230 offer_answer_step() soa_static(0x7ff8b0038c40, > soa_generate_answer): upgrade with remote description > soa_static.c:1264 offer_answer_step() soa_static(0x7ff8b0038c40, > soa_generate_answer): marking rejected media > soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff8d9924960, > 0x7ff8b0b7f8e0, ""): called > soa_static.c:1446 offer_answer_step() soa_static(0x7ff8b0038c40, > soa_generate_answer): storing local description > soa.c:1730 soa_activate() soa_activate(static::0x7ff8b0038c40, (nil)) > called > soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7ff8b0038c40, > [(nil)], [0x7ff8d9926ad0], [0x7ff8d9926acc]) called > tport.c:3257 tport_tsend() tport_tsend(0x7ff8b0004930) tpn = UDP/ > 5.2.2.6:5060 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 5.2.2.6:506 > > > > > > Regards > Raman > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Mon Oct 9 08:48:56 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Mon, 9 Oct 2017 08:48:56 +0000 Subject: [Freeswitch-users] bad ringback tone if I use OPUS Message-ID: <93876bb54dca4ac6af1b94ae184744dc@c4b.de> Hi All, for outbound calls I have configured following two lines in the dialplan: For the G711 and the G722 Codecs the ringback works fine but with the OPUS codec there is a bad noice after the first ringback tone hearable. This problem is veryfied with Bria 4 and the ICE Link Stack Have anyone an idea? Thanks a lot! -------------- next part -------------- An HTML attachment was scrubbed... URL: From josedavid at zennio.com Mon Oct 9 12:07:32 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Mon, 9 Oct 2017 14:07:32 +0200 Subject: [Freeswitch-users] Calls only between members of the same group Message-ID: Hi, I would like to restrict all calls so that they can only be made between users in the same group (there may be many groups) but I have not found anything in particular. I have been playing with the parameters "toll_allow" and "callgroup" but still calling between members of different groups. Does anyone know how to do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Mon Oct 9 13:22:33 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 9 Oct 2017 10:22:33 -0300 Subject: [Freeswitch-users] Calls only between members of the same group In-Reply-To: References: Message-ID: Use different contexts. On Mon, Oct 9, 2017 at 9:07 AM, Jose David Jurado Alonso < josedavid at zennio.com> wrote: > Hi, > > I would like to restrict all calls so that they can only be made between > users in the same group (there may be many groups) but I have not found > anything in particular. > > I have been playing with the parameters "toll_allow" and "callgroup" but > still calling between members of different groups. > > Does anyone know how to do this? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos.oancea at vonage.com Mon Oct 9 13:35:36 2017 From: dragos.oancea at vonage.com (Oancea, Dragos) Date: Mon, 9 Oct 2017 14:35:36 +0100 Subject: [Freeswitch-users] bad ringback tone if I use OPUS In-Reply-To: <93876bb54dca4ac6af1b94ae184744dc@c4b.de> References: <93876bb54dca4ac6af1b94ae184744dc@c4b.de> Message-ID: Hi, First try to see if it happens with other VBR codec and then please open a jira and add logs and network traces. Regards, Dragos On Mon, Oct 9, 2017 at 9:48 AM, Alexander Haugg wrote: > Hi All, > > > > for outbound calls I have configured following two lines in the dialplan: > > > > > > > > > For the G711 and the G722 Codecs the ringback works fine but with the OPUS > codec there is a bad noice after the first ringback tone hearable. > > This problem is veryfied with Bria 4 and the ICE Link Stack > > > > Have anyone an idea? > > > > Thanks a lot! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 mike at jerris.com Mon Oct 9 17:41:34 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Oct 2017 13:41:34 -0400 Subject: [Freeswitch-users] RRNoise - would be pretty cool to see a plugin for this in FS such as for the conference module In-Reply-To: References: Message-ID: <42DCF84B-B18F-4C01-98A1-33E6D1547F97@jerris.com> stuff like this is MUCH better when done at the endpoint, not in the network. That being said, its probably doable if someone wanted to make it happen. > On Oct 1, 2017, at 5:52 PM, Nathan Neulinger wrote: > > https://tech.slashdot.org/story/17/10/01/2014216/donate-you-noise-to-xiphmozillas-deep-learning-noise-suppression-project > > https://people.xiph.org/~jm/demo/rnnoise/ > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect From mike at jerris.com Mon Oct 9 17:53:00 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Oct 2017 13:53:00 -0400 Subject: [Freeswitch-users] Any way to configure failover for shoutcast moh streams? In-Reply-To: References: Message-ID: <4F4D11DE-F8FA-4722-8D58-4CDFC230B52B@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/mod_local_stream#mod_local_stream-Usingaplaylist > On Oct 2, 2017, at 10:49 AM, Nathan Neulinger wrote: > > I know with the plain file MOH you can point to a directory and have it shuffle - is there any way to do that with streams or to have it fail over from a stream to a static file? > > -- Nathan > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Oct 9 18:13:04 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Oct 2017 14:13:04 -0400 Subject: [Freeswitch-users] PLAYBACK_STOP event not sent trhough ESL for playback if speed has been changed with uuid_fileman In-Reply-To: References: Message-ID: <9E0E305C-40CD-428E-B814-C398B8A5D08B@jerris.com> sounds like a bug. https://freeswitch.org/jira > On Oct 3, 2017, at 6:21 AM, Harangozó, László wrote: > > Hi guys, > > Setup: > Inbound calls are managed in a separate app through an ESL connection, so the FS dialplan is simple: > > > > > > > > > If a playback operation is started (ESL ExecuteApplication playback) for such a call then both PLAYBACK_START and PLAYBACK_STOP events are sent for the playback application. > Except for the case when during play the speed is changed, for example with the following bgapi call: > uuid_fileman 8bca0318-a5f5-4170-b82a-d7ab09d018d4 speed:+1 > > When the speed is changed in such a way, the PLAYBACK_STOP event won't be sent. > > Environment: > Platform: Windows > Branch: v1.6 > File being played: Issue present with all sound files I have tried > > The issue happens in a consistent way (all the time when the described scenario happens) > > Does anyone have a clue, how to get FS to send the PLAYBACK_STOP events? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Oct 9 18:30:30 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Oct 2017 14:30:30 -0400 Subject: [Freeswitch-users] session id (rfc7329) support (VoIP and Verto)? In-Reply-To: References: Message-ID: <30ED04C7-7D9C-4FF3-9FD9-6B90A2669BAC@jerris.com> We do not, and most certainly do not in Verto as this is a SIP spec. > On Oct 6, 2017, at 1:50 PM, Mundkowsky, Robert wrote: > > Does FreeSWITCH have session id (rfc7329) support (VoIP and Verto)? > > > Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Oct 9 18:34:19 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 9 Oct 2017 14:34:19 -0400 Subject: [Freeswitch-users] Issue with database updates In-Reply-To: <7197041B-AD3C-4018-AB27-8B7B77D3B45D@magicmail.mooo.com> References: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> <959A217B-65B9-4413-B12E-C6A880F68334@vallimamod.org> <7197041B-AD3C-4018-AB27-8B7B77D3B45D@magicmail.mooo.com> Message-ID: it updates them with the values in channel vars. > On Oct 8, 2017, at 2:13 PM, Rick Jarvis wrote: > > I had resorted to updating the table directly through SQL as FS doesn't appear to dynamically update the table if I use uuid_setvar :/ > >> On 8 Oct 2017, at 19:05, Vallimamod Abdullah > wrote: >> >> Hi, >> >> I am not sure how you are updating these columns but if you are doing it directly through sql, I don't think it's the correct way. >> To set or update a value on a "presence data" column, you have to define or update a channel variable with the same name as the column. If there is no variable defined, the corresponding column will be set to NULL on the state change. >> >> Hope this helps. >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sipsolutions.fr >> tel: +33 6 62 60 68 97 >> . >> >>> On 8 Oct 2017, at 16:50, Rick Jarvis > wrote: >>> >>> Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols ) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Mon Oct 9 18:38:38 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Mon, 9 Oct 2017 19:38:38 +0100 Subject: [Freeswitch-users] Issue with database updates In-Reply-To: References: <2D37FE45-0C09-425F-A77A-4B0D2DD91CE2@magicmail.mooo.com> <959A217B-65B9-4413-B12E-C6A880F68334@vallimamod.org> <7197041B-AD3C-4018-AB27-8B7B77D3B45D@magicmail.mooo.com> Message-ID: Sure, but surely the point of SQL is to update only the affected fields? If it’s going to read the values out one by one, and then set them one by one on a following transaction, it just creates the opportunity for stale data, which is what I’m getting :( > On 9 Oct 2017, at 19:34, Michael Jerris wrote: > > it updates them with the values in channel vars. > >> On Oct 8, 2017, at 2:13 PM, Rick Jarvis > wrote: >> >> I had resorted to updating the table directly through SQL as FS doesn't appear to dynamically update the table if I use uuid_setvar :/ >> >>> On 8 Oct 2017, at 19:05, Vallimamod Abdullah > wrote: >>> >>> Hi, >>> >>> I am not sure how you are updating these columns but if you are doing it directly through sql, I don't think it's the correct way. >>> To set or update a value on a "presence data" column, you have to define or update a channel variable with the same name as the column. If there is no variable defined, the corresponding column will be set to NULL on the state change. >>> >>> Hope this helps. >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sipsolutions.fr >>> tel: +33 6 62 60 68 97 >>> . >>> >>>> On 8 Oct 2017, at 16:50, Rick Jarvis > wrote: >>>> >>>> Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data+cols ) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!? >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 shakumarsoftware at gmail.com Mon Oct 9 20:52:18 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Mon, 9 Oct 2017 14:52:18 -0600 Subject: [Freeswitch-users] Auth and DNS load balancing. Message-ID: Hello, I have the below scenario. FS talking to a service provider and the sip server has DNS load balancing. (FS) INVITE --> abc.com( ip = 10.10.10.10) 407 <------abc.com Ack-----------> abc.com INVITE(credentials)-->abc.com (ip = 11.11.11.11) 407<-------------------abc.com Ack------------> abc.com The session is destroyed at this point and Freeswitch does not resend the INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. The sofia traces indicate session terminated [904]. I understand that if DNS is going to load balance for every INVITE then the call will never be established. But I still would like to see if it can be done? Anyone else run into this issue ? thank you, Shaks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Oct 9 22:04:26 2017 From: brian at freeswitch.com (Brian West) Date: Mon, 9 Oct 2017 17:04:26 -0500 Subject: [Freeswitch-users] Auth and DNS load balancing. In-Reply-To: References: Message-ID: No way you can fix this, what is the TTL on your DNS records? ./b On Mon, Oct 9, 2017 at 3:52 PM, Sharath Kumar wrote: > Hello, > I have the below scenario. > > FS talking to a service provider and the sip server has DNS load > balancing. > > (FS) > INVITE --> abc.com( ip = 10.10.10.10) > 407 <------abc.com > Ack-----------> abc.com > > INVITE(credentials)-->abc.com (ip = 11.11.11.11) > 407<-------------------abc.com > Ack------------> abc.com > > The session is destroyed at this point and Freeswitch does not resend the > INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. > > The sofia traces indicate session terminated [904]. I understand that if > DNS is going to load balance for every INVITE then the call will never be > established. But I still would like to see if it can be done? Anyone else > run into this issue ? > > thank you, > Shaks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.com *Twitter: @FreeSWITCH , @cluecon* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) -------------- next part -------------- An HTML attachment was scrubbed... URL: From rfmundkowsky at yahoo.com Mon Oct 9 22:16:51 2017 From: rfmundkowsky at yahoo.com (Robert Mundkowsky) Date: Mon, 09 Oct 2017 18:16:51 -0400 Subject: [Freeswitch-users] session id (rfc7329) support (VoIP and Verto)? Message-ID: <3x4hf728onwhpmv5ydf220cb.1507587008977@email.android.com> Maybe a better question is there a different mechanism for tracking a SIP based call that orginated from Verto across multiple SIP based systems?  I suppose Homer can be used for this, but seems like the overkill solution. -------- Original message -------- From: Michael Jerris Date: 10/9/17 2:30 PM (GMT-05:00) To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] session id (rfc7329) support (VoIP and Verto)? We do not, and most certainly do not in Verto as this is a SIP spec. On Oct 6, 2017, at 1:50 PM, Mundkowsky, Robert wrote: Does FreeSWITCH have session id (rfc7329) support (VoIP and Verto)?  Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakumarsoftware at gmail.com Mon Oct 9 22:19:22 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Mon, 9 Oct 2017 16:19:22 -0600 Subject: [Freeswitch-users] Auth and DNS load balancing. In-Reply-To: References: Message-ID: Thanks Brian! The DNS is from the SIP provider end. I have no idea but just from the call failure rates it is in the order of minutes is my guess. On Mon, Oct 9, 2017 at 4:04 PM, Brian West wrote: > No way you can fix this, what is the TTL on your DNS records? > > ./b > > On Mon, Oct 9, 2017 at 3:52 PM, Sharath Kumar > wrote: > >> Hello, >> I have the below scenario. >> >> FS talking to a service provider and the sip server has DNS load >> balancing. >> >> (FS) >> INVITE --> abc.com( ip = 10.10.10.10) >> 407 <------abc.com >> Ack-----------> abc.com >> >> INVITE(credentials)-->abc.com (ip = 11.11.11.11) >> 407<-------------------abc.com >> Ack------------> abc.com >> >> The session is destroyed at this point and Freeswitch does not resend the >> INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. >> >> The sofia traces indicate session terminated [904]. I understand that if >> DNS is going to load balance for every INVITE then the call will never be >> established. But I still would like to see if it can be done? Anyone else >> run into this issue ? >> >> thank you, >> Shaks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.com > > *Twitter: @FreeSWITCH , @cluecon* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 brians at iptel.co Tue Oct 10 07:33:33 2017 From: brians at iptel.co (Brian :) Date: Tue, 10 Oct 2017 08:33:33 +0100 Subject: [Freeswitch-users] Auth and DNS load balancing. In-Reply-To: References: Message-ID: This providers behaviour is broken. Stick IP address as proxy in GW definition - fs will route calls to the IP - you can setup a second GW with their other IP and failover to that in the event that GW1 is down. On Mon, Oct 9, 2017 at 11:19 PM, Sharath Kumar wrote: > Thanks Brian! The DNS is from the SIP provider end. I have no idea but just > from the call failure rates it is in the order of minutes is my guess. > > > On Mon, Oct 9, 2017 at 4:04 PM, Brian West wrote: >> >> No way you can fix this, what is the TTL on your DNS records? >> >> ./b >> >> On Mon, Oct 9, 2017 at 3:52 PM, Sharath Kumar >> wrote: >>> >>> Hello, >>> I have the below scenario. >>> >>> FS talking to a service provider and the sip server has DNS load >>> balancing. >>> >>> (FS) >>> INVITE --> abc.com( ip = 10.10.10.10) >>> 407 <------abc.com >>> Ack-----------> abc.com >>> >>> INVITE(credentials)-->abc.com (ip = 11.11.11.11) >>> 407<-------------------abc.com >>> Ack------------> abc.com >>> >>> The session is destroyed at this point and Freeswitch does not resend the >>> INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. >>> >>> The sofia traces indicate session terminated [904]. I understand that if >>> DNS is going to load balance for every INVITE then the call will never be >>> established. But I still would like to see if it can be done? Anyone else >>> run into this issue ? >>> >>> thank you, >>> Shaks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> >> Brian West >> brian at freeswitch.com >> >> Twitter: @FreeSWITCH , @cluecon >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> >> Got Bugs? Report them here! | Reddit: /r/freeswitch >> >> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From akshaya18j at gmail.com Tue Oct 10 09:40:43 2017 From: akshaya18j at gmail.com (Akshaya j) Date: Tue, 10 Oct 2017 15:10:43 +0530 Subject: [Freeswitch-users] [ERR] switch_core_video.c:2879 This function is not available, libpng not installed Message-ID: Hi team.! In Centos7 with libpng 1.5 version When I dial conference number I can witness an ERR [ERR] switch_core_video.c:2879 This function is not available, libpng not installed... Conference call doesn't use video its just audio call.. The calls works properly but this error bothers me of what the problem is? Someone please help in fixing this. I run another FS setup in Debian machine where there s no such error as libpng 1.6 version is available Note: libpng is up to date in my centos 7(libpng1.5 which is what available for centos7) but freeswitch needs libpng 1.6, I guess. Thanks in advance Akshaya J -------------- next part -------------- An HTML attachment was scrubbed... URL: From ovoshlook at gmail.com Tue Oct 10 09:43:32 2017 From: ovoshlook at gmail.com (Yuriy Gorlichenko) Date: Tue, 10 Oct 2017 12:43:32 +0300 Subject: [Freeswitch-users] set variable at te chatplan with Lua Message-ID: Hi In dialplan i can set veriable via session:execute ("set",) or session:setVariable (,) In chat plan exists only "set" application but in can not bee executed via message (analog of session) how to set variables here? I tried to use api:chat({} proto|sender|receiver|message) But it is does not work I exepted because of chat function description but tried anywhay. -------------- next part -------------- An HTML attachment was scrubbed... URL: From avi at avimarcus.net Tue Oct 10 09:54:47 2017 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 10 Oct 2017 09:54:47 +0000 Subject: [Freeswitch-users] Stats collection (telegraf), specifically: all limit hash results Message-ID: <0100015f05b548ef-6e7644b4-ff91-4089-9e2f-034ffe11a191-000000@email.amazonses.com> I see an API command to query a specific limit. When using db back end, I could query the db. Is there a way to get the whole hash back end in one shot? I only see `hash select/realm/key` in the docs. I'm looking at integrating with telegraf data collector. *Previous art for collectors:* Active channels/calls/cps here (the common ones): https://github.com/areski/freeswitch-telegraf-plugin and also active calls per profile + some history here: https://github.com/moises-silva/freeswitch-telegraf Reinman adds conference list: https://github.com/riemann/riemann-tools/blob/master/bin/riemann-freeswitch datadog: adds skipped packets, g729 channel count, events of reloadxml/modules https://github.com/wimactel/FreeSwitch-DataDog-Metrics/blob/master/metrics.py statsd: adds registration count https://github.com/sangoma/mod_statsd prometheus with registrations and channels, but custom counters instead of accesing limit: https://github.com/moises-silva/mod_prometheus -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Mon Oct 9 22:07:58 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 9 Oct 2017 18:07:58 -0400 Subject: [Freeswitch-users] Auth and DNS load balancing. In-Reply-To: References: Message-ID: You should read how the digest authentication works. Then you will realize this, as you want won't work. Google smart DNS for freeswitch You will find good stuff that will resolve your issue in a different approach. On Oct 9, 2017 4:53 PM, "Sharath Kumar" wrote: > Hello, > I have the below scenario. > > FS talking to a service provider and the sip server has DNS load > balancing. > > (FS) > INVITE --> abc.com( ip = 10.10.10.10) > 407 <------abc.com > Ack-----------> abc.com > > INVITE(credentials)-->abc.com (ip = 11.11.11.11) > 407<-------------------abc.com > Ack------------> abc.com > > The session is destroyed at this point and Freeswitch does not resend the > INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. > > The sofia traces indicate session terminated [904]. I understand that if > DNS is going to load balance for every INVITE then the call will never be > established. But I still would like to see if it can be done? Anyone else > run into this issue ? > > thank you, > Shaks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 rmundkowsky at ets.org Mon Oct 9 14:38:38 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Mon, 9 Oct 2017 14:38:38 +0000 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: I am curious, what type of situation causes this problem? I mean, we use Verto for a conference, granted without Nginx, and seems like there are always packets being sent back and forth even when there is no audio or video. Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Friday, October 6, 2017 4:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sending keepalives on Verto websocket There are various suggested solutions out there. On the Nginx config side, easiest is to increase the proxy_read_timeout setting to something high, but that seems like it could result in connections getting hung open if the server process goes away. Then there’s this module which looks pretty fancy and claims to solve the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module — it requires custom compiling nginx though… I think I’ll stick with my current solution, which basically follows the other suggestion of having the server send periodic keepalives, except that I’m initiating from the client. Steps for others interested: 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant domains in your user directory 2. Do something like this after a successful websocket connection: MODULE.keepAliveTimer = setInterval(function() { verto.rpcClient.call("echo", {keepalive: true}); }, VERTO_KEEPALIVE_INTERVAL) 3. Add some cleanup on websocket close: if (MODULE.keepAliveTimer) { clearInterval(MODULE.keepAliveTimer); MODULE.keepAliveTimer = null; } Request and response lengths are both under 150 characters, so pretty short, and I set the intervals at 50 seconds, since Nginx’s default timeout is 60 seconds. On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert > wrote: Have you checked if Nginx has timeout settings and/or keep alive settings? https://stackoverflow.com/questions/10550558/nginx-tcp-websockets-timeout-keepalive-config Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Thursday, October 5, 2017 11:07 PM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Sending keepalives on Verto websocket I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality? ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Oct 10 19:26:13 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 10 Oct 2017 15:26:13 -0400 Subject: [Freeswitch-users] [ERR] switch_core_video.c:2879 This function is not available, libpng not installed In-Reply-To: References: Message-ID: <2506C165-6031-4642-887A-1ECA007C8272@jerris.com> its trying to read a png file for something. are you sure you don’t have any of the banner png’s or mute images or avatar stuff configured? that being said, we should still work w/ libpng 1.5, just don’t support alpha channel if you are using the version of libpng that doesn’t support it. > On Oct 10, 2017, at 5:40 AM, Akshaya j wrote: > > Hi team.! > In Centos7 with libpng 1.5 version > When I dial conference number I can witness an ERR > [ERR] switch_core_video.c:2879 This function is not available, libpng not installed... > Conference call doesn't use video its just audio call.. The calls works properly but this error bothers me of what the problem is? > Someone please help in fixing this. > I run another FS setup in Debian machine where there s no such error as libpng 1.6 version is available > Note: > libpng is up to date in my centos 7(libpng1.5 which is what available for centos7) but freeswitch needs libpng 1.6, I guess. > > Thanks in advance > Akshaya J > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 chad at apartmentlines.com Tue Oct 10 23:51:39 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Tue, 10 Oct 2017 16:51:39 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: I’ve seen this issue when everyone is audio muted and/or doesn’t trigger a talking event for more than 60 seconds, which in my use case does happen. I would agree under normal videoconferencing circumstances it would be rare to go 60 seconds without a single event being sent along the websocket. On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert wrote: > I am curious, what type of situation causes this problem? I mean, we use > Verto for a conference, granted without Nginx, and seems like there are > always packets being sent back and forth even when there is no audio or > video. > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Friday, October 6, 2017 4:53 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket > > > > There are various suggested solutions out there. On the Nginx config side, > easiest is to increase the proxy_read_timeout setting to something high, > but that seems like it could result in connections getting hung open if the > server process goes away. > > > > Then there’s this module which looks pretty fancy and claims to solve the > problem: https://github.com/yaoweibin/nginx_tcp_proxy_module > > — it requires custom compiling nginx though… > > > > I think I’ll stick with my current solution, which basically follows the > other suggestion of having the server send periodic keepalives, except that > I’m initiating from the client. Steps for others interested: > > > > 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant > domains in your user directory > > > > 2. Do something like this after a successful websocket connection: > > > > MODULE.keepAliveTimer = setInterval(function() { > > verto.rpcClient.call("echo", {keepalive: true}); > > }, VERTO_KEEPALIVE_INTERVAL) > > > > 3. Add some cleanup on websocket close: > > > > if (MODULE.keepAliveTimer) { > > clearInterval(MODULE.keepAliveTimer); > > MODULE.keepAliveTimer = null; > > } > > > > Request and response lengths are both under 150 characters, so pretty > short, and I set the intervals at 50 seconds, since Nginx’s default timeout > is 60 seconds. > > > > On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert > wrote: > > Have you checked if Nginx has timeout settings and/or keep alive settings? > > > > https://stackoverflow.com/questions/10550558/nginx-tcp- > websockets-timeout-keepalive-config > > > > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Thursday, October 5, 2017 11:07 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket > > > > I've switched to using Nginx to proxy Verto websockets, and have run into > a small snag: by default, if Nginx doesn't read any data from a proxy > backend within 60 seconds, it closes the connection, even for websockets. > > > > It appears the recommended solution is to have the server send some kind > of regular keepalive. I poked around in mod_verto.c and found a > 'request.keepalive' variable, but I'm unclear how to set that in the > request, and/or if it even accomplishes what I'm wanting. > > > > I've solved the issue for now by periodically sending a JSON RPC 'echo' > request along the websocket every 50 seconds, and the reply from the server > is enough to keep the connection open. This is fine, but I am curious if > there's a way to do it just from the server side, and if not, if it's worth > it to add a setting to enable that functionality? > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > > > Thank you for your compliance. > ------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 italo at freeswitch.org Wed Oct 11 02:02:08 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 10 Oct 2017 23:02:08 -0300 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Are we talking about proxy_read_timeout param? On Tue, Oct 10, 2017 at 8:51 PM, Chad Phillips wrote: > I’ve seen this issue when everyone is audio muted and/or doesn’t trigger a > talking event for more than 60 seconds, which in my use case does happen. I > would agree under normal videoconferencing circumstances it would be rare > to go 60 seconds without a single event being sent along the websocket. > > On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert > wrote: > >> I am curious, what type of situation causes this problem? I mean, we use >> Verto for a conference, granted without Nginx, and seems like there are >> always packets being sent back and forth even when there is no audio or >> video. >> >> >> >> Robert >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Chad Phillips >> *Sent:* Friday, October 6, 2017 4:53 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket >> >> >> >> There are various suggested solutions out there. On the Nginx config >> side, easiest is to increase the proxy_read_timeout setting to something >> high, but that seems like it could result in connections getting hung open >> if the server process goes away. >> >> >> >> Then there’s this module which looks pretty fancy and claims to solve the >> problem: https://github.com/yaoweibin/nginx_tcp_proxy_module >> >> — it requires custom compiling nginx though… >> >> >> >> I think I’ll stick with my current solution, which basically follows the >> other suggestion of having the server send periodic keepalives, except that >> I’m initiating from the client. Steps for others interested: >> >> >> >> 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant >> domains in your user directory >> >> >> >> 2. Do something like this after a successful websocket connection: >> >> >> >> MODULE.keepAliveTimer = setInterval(function() { >> >> verto.rpcClient.call("echo", {keepalive: true}); >> >> }, VERTO_KEEPALIVE_INTERVAL) >> >> >> >> 3. Add some cleanup on websocket close: >> >> >> >> if (MODULE.keepAliveTimer) { >> >> clearInterval(MODULE.keepAliveTimer); >> >> MODULE.keepAliveTimer = null; >> >> } >> >> >> >> Request and response lengths are both under 150 characters, so pretty >> short, and I set the intervals at 50 seconds, since Nginx’s default timeout >> is 60 seconds. >> >> >> >> On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert >> wrote: >> >> Have you checked if Nginx has timeout settings and/or keep alive settings? >> >> >> >> https://stackoverflow.com/questions/10550558/nginx-tcp-webso >> ckets-timeout-keepalive-config >> >> >> >> >> >> >> Robert >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Chad Phillips >> *Sent:* Thursday, October 5, 2017 11:07 PM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket >> >> >> >> I've switched to using Nginx to proxy Verto websockets, and have run into >> a small snag: by default, if Nginx doesn't read any data from a proxy >> backend within 60 seconds, it closes the connection, even for websockets. >> >> >> >> It appears the recommended solution is to have the server send some kind >> of regular keepalive. I poked around in mod_verto.c and found a >> 'request.keepalive' variable, but I'm unclear how to set that in the >> request, and/or if it even accomplishes what I'm wanting. >> >> >> >> I've solved the issue for now by periodically sending a JSON RPC 'echo' >> request along the websocket every 50 seconds, and the reply from the server >> is enough to keep the connection open. This is fine, but I am curious if >> there's a way to do it just from the server side, and if not, if it's worth >> it to add a setting to enable that functionality? >> >> >> ------------------------------ >> >> This e-mail and any files transmitted with it may contain privileged or >> confidential information. It is solely for use by the individual for whom >> it is intended, even if addressed incorrectly. If you received this e-mail >> in error, please notify the sender; do not disclose, copy, distribute, or >> take any action in reliance on the contents of this information; and delete >> it from your system. Any other use of this e-mail is prohibited. >> >> >> >> Thank you for your compliance. >> ------------------------------ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> ------------------------------ >> >> This e-mail and any files transmitted with it may contain privileged or >> confidential information. It is solely for use by the individual for whom >> it is intended, even if addressed incorrectly. If you received this e-mail >> in error, please notify the sender; do not disclose, copy, distribute, or >> take any action in reliance on the contents of this information; and delete >> it from your system. Any other use of this e-mail is prohibited. >> >> Thank you for your compliance. >> ------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Wed Oct 11 03:42:36 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Tue, 10 Oct 2017 20:42:36 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: proxy_read_timeout would be an approach to solve the issue from the Nginx config side. I don’t like that solution as much, though, because it requires guessing how long the socket might be idle, whereas the keepalives can be reliably sent to always prevent the timeout. On Tue, Oct 10, 2017 at 7:02 PM, Ítalo Rossi wrote: > Are we talking about proxy_read_timeout param? > > On Tue, Oct 10, 2017 at 8:51 PM, Chad Phillips > wrote: > >> I’ve seen this issue when everyone is audio muted and/or doesn’t trigger >> a talking event for more than 60 seconds, which in my use case does happen. >> I would agree under normal videoconferencing circumstances it would be rare >> to go 60 seconds without a single event being sent along the websocket. >> >> On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert >> wrote: >> >>> I am curious, what type of situation causes this problem? I mean, we >>> use Verto for a conference, granted without Nginx, and seems like there are >>> always packets being sent back and forth even when there is no audio or >>> video. >>> >>> >>> >>> Robert >>> >>> >>> >>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >>> es at lists.freeswitch.org] *On Behalf Of *Chad Phillips >>> *Sent:* Friday, October 6, 2017 4:53 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket >>> >>> >>> >>> There are various suggested solutions out there. On the Nginx config >>> side, easiest is to increase the proxy_read_timeout setting to >>> something high, but that seems like it could result in connections getting >>> hung open if the server process goes away. >>> >>> >>> >>> Then there’s this module which looks pretty fancy and claims to solve >>> the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module >>> >>> — it requires custom compiling nginx though… >>> >>> >>> >>> I think I’ll stick with my current solution, which basically follows the >>> other suggestion of having the server send periodic keepalives, except that >>> I’m initiating from the client. Steps for others interested: >>> >>> >>> >>> 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant >>> domains in your user directory >>> >>> >>> >>> 2. Do something like this after a successful websocket connection: >>> >>> >>> >>> MODULE.keepAliveTimer = setInterval(function() { >>> >>> verto.rpcClient.call("echo", {keepalive: true}); >>> >>> }, VERTO_KEEPALIVE_INTERVAL) >>> >>> >>> >>> 3. Add some cleanup on websocket close: >>> >>> >>> >>> if (MODULE.keepAliveTimer) { >>> >>> clearInterval(MODULE.keepAliveTimer); >>> >>> MODULE.keepAliveTimer = null; >>> >>> } >>> >>> >>> >>> Request and response lengths are both under 150 characters, so pretty >>> short, and I set the intervals at 50 seconds, since Nginx’s default timeout >>> is 60 seconds. >>> >>> >>> >>> On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert >>> wrote: >>> >>> Have you checked if Nginx has timeout settings and/or keep alive >>> settings? >>> >>> >>> >>> https://stackoverflow.com/questions/10550558/nginx-tcp-webso >>> ckets-timeout-keepalive-config >>> >>> >>> >>> >>> >>> >>> Robert >>> >>> >>> >>> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >>> es at lists.freeswitch.org] *On Behalf Of *Chad Phillips >>> *Sent:* Thursday, October 5, 2017 11:07 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket >>> >>> >>> >>> I've switched to using Nginx to proxy Verto websockets, and have run >>> into a small snag: by default, if Nginx doesn't read any data from a proxy >>> backend within 60 seconds, it closes the connection, even for websockets. >>> >>> >>> >>> It appears the recommended solution is to have the server send some kind >>> of regular keepalive. I poked around in mod_verto.c and found a >>> 'request.keepalive' variable, but I'm unclear how to set that in the >>> request, and/or if it even accomplishes what I'm wanting. >>> >>> >>> >>> I've solved the issue for now by periodically sending a JSON RPC 'echo' >>> request along the websocket every 50 seconds, and the reply from the server >>> is enough to keep the connection open. This is fine, but I am curious if >>> there's a way to do it just from the server side, and if not, if it's worth >>> it to add a setting to enable that functionality? >>> >>> >>> ------------------------------ >>> >>> This e-mail and any files transmitted with it may contain privileged or >>> confidential information. It is solely for use by the individual for whom >>> it is intended, even if addressed incorrectly. If you received this e-mail >>> in error, please notify the sender; do not disclose, copy, distribute, or >>> take any action in reliance on the contents of this information; and delete >>> it from your system. Any other use of this e-mail is prohibited. >>> >>> >>> >>> Thank you for your compliance. >>> ------------------------------ >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> >>> http://confluence.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> ------------------------------ >>> >>> This e-mail and any files transmitted with it may contain privileged or >>> confidential information. It is solely for use by the individual for whom >>> it is intended, even if addressed incorrectly. If you received this e-mail >>> in error, please notify the sender; do not disclose, copy, distribute, or >>> take any action in reliance on the contents of this information; and delete >>> it from your system. Any other use of this e-mail is prohibited. >>> >>> Thank you for your compliance. >>> ------------------------------ >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Wed Oct 11 04:40:17 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Oct 2017 04:40:17 +0000 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: The whole idea of a proxy read timeout is a hack. Working around it forces more hacks. Tcp has this covered...... On Tue, Oct 10, 2017 at 10:43 PM Chad Phillips wrote: > proxy_read_timeout would be an approach to solve the issue from the Nginx > config side. I don’t like that solution as much, though, because it > requires guessing how long the socket might be idle, whereas the keepalives > can be reliably sent to always prevent the timeout. > > On Tue, Oct 10, 2017 at 7:02 PM, Ítalo Rossi wrote: > >> Are we talking about proxy_read_timeout param? >> >> On Tue, Oct 10, 2017 at 8:51 PM, Chad Phillips >> wrote: >> >>> I’ve seen this issue when everyone is audio muted and/or doesn’t trigger >>> a talking event for more than 60 seconds, which in my use case does happen. >>> I would agree under normal videoconferencing circumstances it would be rare >>> to go 60 seconds without a single event being sent along the websocket. >>> >>> On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert >>> wrote: >>> >>>> I am curious, what type of situation causes this problem? I mean, we >>>> use Verto for a conference, granted without Nginx, and seems like there are >>>> always packets being sent back and forth even when there is no audio or >>>> video. >>>> >>>> >>>> >>>> Robert >>>> >>>> >>>> >>>> *From:* FreeSWITCH-users [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chad >>>> Phillips >>>> *Sent:* Friday, October 6, 2017 4:53 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket >>>> >>>> >>>> >>>> There are various suggested solutions out there. On the Nginx config >>>> side, easiest is to increase the proxy_read_timeout setting to >>>> something high, but that seems like it could result in connections getting >>>> hung open if the server process goes away. >>>> >>>> >>>> >>>> Then there’s this module which looks pretty fancy and claims to solve >>>> the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module >>>> >>>> — it requires custom compiling nginx though… >>>> >>>> >>>> >>>> I think I’ll stick with my current solution, which basically follows >>>> the other suggestion of having the server send periodic keepalives, except >>>> that I’m initiating from the client. Steps for others interested: >>>> >>>> >>>> >>>> 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant >>>> domains in your user directory >>>> >>>> >>>> >>>> 2. Do something like this after a successful websocket connection: >>>> >>>> >>>> >>>> MODULE.keepAliveTimer = setInterval(function() { >>>> >>>> verto.rpcClient.call("echo", {keepalive: true}); >>>> >>>> }, VERTO_KEEPALIVE_INTERVAL) >>>> >>>> >>>> >>>> 3. Add some cleanup on websocket close: >>>> >>>> >>>> >>>> if (MODULE.keepAliveTimer) { >>>> >>>> clearInterval(MODULE.keepAliveTimer); >>>> >>>> MODULE.keepAliveTimer = null; >>>> >>>> } >>>> >>>> >>>> >>>> Request and response lengths are both under 150 characters, so pretty >>>> short, and I set the intervals at 50 seconds, since Nginx’s default timeout >>>> is 60 seconds. >>>> >>>> >>>> >>>> On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert >>>> wrote: >>>> >>>> Have you checked if Nginx has timeout settings and/or keep alive >>>> settings? >>>> >>>> >>>> >>>> >>>> https://stackoverflow.com/questions/10550558/nginx-tcp-websockets-timeout-keepalive-config >>>> >>>> >>>> >>>> >>>> >>>> >>>> Robert >>>> >>>> >>>> >>>> *From:* FreeSWITCH-users [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chad >>>> Phillips >>>> *Sent:* Thursday, October 5, 2017 11:07 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket >>>> >>>> >>>> >>>> I've switched to using Nginx to proxy Verto websockets, and have run >>>> into a small snag: by default, if Nginx doesn't read any data from a proxy >>>> backend within 60 seconds, it closes the connection, even for websockets. >>>> >>>> >>>> >>>> It appears the recommended solution is to have the server send some >>>> kind of regular keepalive. I poked around in mod_verto.c and found a >>>> 'request.keepalive' variable, but I'm unclear how to set that in the >>>> request, and/or if it even accomplishes what I'm wanting. >>>> >>>> >>>> >>>> I've solved the issue for now by periodically sending a JSON RPC 'echo' >>>> request along the websocket every 50 seconds, and the reply from the server >>>> is enough to keep the connection open. This is fine, but I am curious if >>>> there's a way to do it just from the server side, and if not, if it's worth >>>> it to add a setting to enable that functionality? >>>> >>>> >>>> ------------------------------ >>>> >>>> This e-mail and any files transmitted with it may contain privileged or >>>> confidential information. It is solely for use by the individual for whom >>>> it is intended, even if addressed incorrectly. If you received this e-mail >>>> in error, please notify the sender; do not disclose, copy, distribute, or >>>> take any action in reliance on the contents of this information; and delete >>>> it from your system. Any other use of this e-mail is prohibited. >>>> >>>> >>>> >>>> Thank you for your compliance. >>>> ------------------------------ >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> >>>> http://confluence.freeswitch.org >>>> >>>> http://www.cluecon.com >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> ------------------------------ >>>> >>>> This e-mail and any files transmitted with it may contain privileged or >>>> confidential information. It is solely for use by the individual for whom >>>> it is intended, even if addressed incorrectly. If you received this e-mail >>>> in error, please notify the sender; do not disclose, copy, distribute, or >>>> take any action in reliance on the contents of this information; and delete >>>> it from your system. Any other use of this e-mail is prohibited. >>>> >>>> Thank you for your compliance. >>>> ------------------------------ >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Ítalo Rossi >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From dayton at voxter.ca Tue Oct 10 21:38:01 2017 From: dayton at voxter.ca (Dayton Turner) Date: Tue, 10 Oct 2017 14:38:01 -0700 Subject: [Freeswitch-users] Choppy and fragmented return-audio problem using Unicast In-Reply-To: <2D82A1C0-6836-4155-89FF-A22341CF325F@vallimamod.org> References: <59D7DF2C.3010904@voxter.com> <2D82A1C0-6836-4155-89FF-A22341CF325F@vallimamod.org> Message-ID: <59DD3DB9.3070501@voxter.com> Hi! I thought that first as well - however we *are* specifying the native flag, and the received audio is definitely PCMU. When I capture it bit-for-bit, and import it into audacity, selecting 8khz mono u-Law makes the audio play back perfectly. I DO see the "Raw Codec Activation Success L16@" in the logs, despite the received audio stream being PCMU, and despite specifying the native flag. We've also attempted streaming L16 back instead of PCMU and its completely incomprehensible static. Whereas now, you can hear it properly, its just stuttered like mad. So, because of this I believe the codec is correct. Dayton Vallimamod Abdullah wrote: > Hi, > > I think you are getting a codec mismatch. Here is how the unicast command works according to the source code: > - If native flag is not set, fs will use the L16 codec to exchange media beetween the session and the unicast socket. You can search for "Raw Codec Activation Success L16@" in the debug log to confirm it. > - If the native flag is set, fs will use the same read and write codec as your current session: the rtp buffer is sent as-is to the unicast socket and the data received from the unicast socket is also sent back as-is to the session. > > Hope this helps! > > Best Regards, From fs at teamviewer.com Wed Oct 11 07:16:07 2017 From: fs at teamviewer.com (Freeswitch) Date: Wed, 11 Oct 2017 07:16:07 +0000 Subject: [Freeswitch-users] Channel Event Issue Message-ID: <8ff56b8f3563479686cdf857beb73b01@Exchange-01.int.rossmanith.com> Hi, Yes, bgapi helps to solve the ESL issue. ESL does not block any more. However, the Channel still blocks when trying to leave the conference. During FreeSWITCH shutdown, the following issue is logged: 2017-10-09 09:42:51.115212 [NOTICE] switch_loadable_module.c:1055 Deleting Application 'conference' 2017-10-09 09:43:01.134791 [ERR] switch_loadable_module.c:1062 Giving up on 'conference' waiting for existing references. 2017-10-09 09:43:01.134791 [NOTICE] switch_loadable_module.c:1110 Deleting API Function 'conference' 2017-10-09 09:43:11.155363 [ERR] switch_loadable_module.c:1118 Giving up on 'conference' waiting for existing references. Regards, Tobias Hi, since updating from FreeSWITCH 1.6.8 to FreeSWITCH 1.6.17 (Windows Server 2012 R2), the Event Socket Handler(inbound mode) sporadically stops sending Channel related Event Info. Once in this situation, the Event Socket Handler does not recover until either reset of the ESL connection by the Host that established the ESL connection or reload of mod_event_socket. Shortly before Event Socket connection drops it seems that there is a problem trying to close a channel. Example Log for failure case: Channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e is never closed after leavig the conference. d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149 at dialin [d9be476b-6a9d-4c96-95fb-5d8728bcc08e] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.399989 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149 at dialin to XML[4600 at default] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.419958 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:35.439966 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149 at dialin] has been answered d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [NOTICE] sofia.c:1012 Hangup sofia/external/149 at dialin [CS_EXECUTE] [NORMAL_CLEARING] d9be476b-6a9d-4c96-95fb-5d8728bcc08e 2017-08-29 10:59:55.940144 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING Example Log for regular case: Channel with uuid 982453be-a321-4825-a944-ecfffa715914 is closed correctly after leaving the conference): 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [NOTICE] switch_channel.c:1104 New Channel sofia/external/149 at dialin [982453be-a321-4825-a944-ecfffa715914] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.340132 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context public 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/149 at dialin to XML[4600 at default] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.360126 [INFO] mod_dialplan_xml.c:637 Processing 149 <149>->4600 in context default 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:55:51.380148 [NOTICE] mod_dptools.c:1312 Channel [sofia/external/149 at dialin] has been answered 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] sofia.c:1012 Hangup sofia/external/149 at dialin [CS_EXECUTE] [NORMAL_CLEARING] 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [INFO] conference_loop.c:1469 Channel leaving conference, cause: NORMAL_CLEARING 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1683 Session 9744 (sofia/external/149 at dialin) Ended 982453be-a321-4825-a944-ecfffa715914 2017-08-29 10:56:11.900321 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/149 at dialin [CS_DESTROY] The channel with uuid d9be476b-6a9d-4c96-95fb-5d8728bcc08e will still be displayed at fs_cli on request of "show channels" freeswitch at 1250-0110> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,accountcode,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context d9be476b-6a9d-4c96-95fb-5d8728bcc08e,inbound,2017-08-29 10:59:35,1503997175,sofia/external/149 at dialin,CS_EXECUTE,149,149,dialin,4600,playback,{playBackId=2573}us/announcement.wav,XML,default,L16,8000,128000,PCMA,8000,64000,,1250-0110,,,,ACTIVE,,,,,,,149,149,dialin,4600,XML,public 1 total. However, trying to retrieve the channel data for this channel e.g. "uuid_dump" fails. freeswitch at 1250-0110> uuid_dump d9be476b-6a9d-4c96-95fb-5d8728bcc08e -ERR No such channel! Same issue exists with FreeSWITCH 1.6.19. We are continuously sending api commands to the channel. Could this be a race condition, receiving an api, while the Channel is trying to close ? Would bgapi help ? Why is it not an issue with FreeSWITCH 1.6.8 ? Regards, Tobias -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Oct 11 00:48:34 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 11 Oct 2017 00:48:34 +0000 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: I am surprised given microphones usually pick up low levels of noise and that Nginx would inspect data packets for data. Are you explicitly stopping data being sent based on push to talk events or VAD? Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Tuesday, October 10, 2017 7:52 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sending keepalives on Verto websocket I’ve seen this issue when everyone is audio muted and/or doesn’t trigger a talking event for more than 60 seconds, which in my use case does happen. I would agree under normal videoconferencing circumstances it would be rare to go 60 seconds without a single event being sent along the websocket. On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert > wrote: I am curious, what type of situation causes this problem? I mean, we use Verto for a conference, granted without Nginx, and seems like there are always packets being sent back and forth even when there is no audio or video. Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Friday, October 6, 2017 4:53 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Sending keepalives on Verto websocket There are various suggested solutions out there. On the Nginx config side, easiest is to increase the proxy_read_timeout setting to something high, but that seems like it could result in connections getting hung open if the server process goes away. Then there’s this module which looks pretty fancy and claims to solve the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module — it requires custom compiling nginx though… I think I’ll stick with my current solution, which basically follows the other suggestion of having the server send periodic keepalives, except that I’m initiating from the client. Steps for others interested: 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant domains in your user directory 2. Do something like this after a successful websocket connection: MODULE.keepAliveTimer = setInterval(function() { verto.rpcClient.call("echo", {keepalive: true}); }, VERTO_KEEPALIVE_INTERVAL) 3. Add some cleanup on websocket close: if (MODULE.keepAliveTimer) { clearInterval(MODULE.keepAliveTimer); MODULE.keepAliveTimer = null; } Request and response lengths are both under 150 characters, so pretty short, and I set the intervals at 50 seconds, since Nginx’s default timeout is 60 seconds. On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert > wrote: Have you checked if Nginx has timeout settings and/or keep alive settings? https://stackoverflow.com/questions/10550558/nginx-tcp-websockets-timeout-keepalive-config Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chad Phillips Sent: Thursday, October 5, 2017 11:07 PM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Sending keepalives on Verto websocket I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality? ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From vikas452 at gmail.com Wed Oct 11 06:33:56 2017 From: vikas452 at gmail.com (vikas sharma) Date: Wed, 11 Oct 2017 12:03:56 +0530 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode In-Reply-To: References: Message-ID: Hi Can i get some suggestions to improve the video call quality and reduce packet loss with Freeswitch server. Thanks On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma wrote: > Anything that can help me in this regard?? > > On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma wrote: > >> Hi >> >> I am making a linphone to linphone call using Freeswitch server in >> Defualt mode but i am facing huge RTP loss and the video call is getting >> stuck in between.I have also found that there is a difference between the >> RTP packets received on A Leg and sent from B-leg and vice versa. >> >> Can somebody help me to understand what exactly happens with the RTP >> packets on server and why there is a difference in the RTP count?? >> >> How can we improve the video call quality in moderate network conditions. >> >> Thanks >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vikas452 at gmail.com Wed Oct 11 06:38:42 2017 From: vikas452 at gmail.com (vikas sharma) Date: Wed, 11 Oct 2017 12:08:42 +0530 Subject: [Freeswitch-users] SIP Calling issues with Freeswitch Message-ID: Hi I am making Linphone to Linphone video calls on android phone.But some times call does not reach the callee side although it is successfully registered with the server. I am checking these calls on mobile data network.Some times call is not established but sometimes it works. Can i get some help on this? Is it due to NAT?Any suggestions on server configuration for this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From babak.freeswitch at gmail.com Wed Oct 11 08:09:13 2017 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Wed, 11 Oct 2017 11:39:13 +0330 Subject: [Freeswitch-users] freetdm errors.. Message-ID: Hi I'm using sangoma a200 with freetdm on freeswitch 1.6 stable branch. Sometimes I see errors like this in console: 2017-10-11 10:58:32.669207 [ERR] ftmod_sangoma_isdn_stack_out.c:574 [s2c31][2:16] Received frame of 323 bytes, exceeding max size of 300 bytes Is this a serious problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Wed Oct 11 12:46:19 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 11 Oct 2017 17:46:19 +0500 Subject: [Freeswitch-users] [curl_sendfile][unable to get response] In-Reply-To: References: Message-ID: Thanks for your reply, yes issue got fix. On Sun, Oct 8, 2017 at 12:25 AM, Yuriy Gorlichenko wrote: > You installed lib for lua 5.1 but Freeswitch requires lua 5.2 > you soulh it to install via luarocks for 5.2 > > > 2017-10-07 21:05 GMT+03:00 Bilal Abbasi : > >> Hi, >> I build the lua script and used the cURL module for that( >> https://github.com/Lua-cURL/Lua-cURLv3) >> I am successfully place the POST request using that module, but when i >> use that with in freeswitch using application lua it raise the ERR >> >> 2017-10-07 17:46:47.592484 [ERR] mod_lua.cpp:203 error loading module >> 'lcurl' from file '/usr/local/lib/lua/5.1/lcurl.so': >> >> /usr/local/lib/lua/5.1/lcurl.so: undefined symbol: lua_tointeger >> >> >> I am not sure why its raising this error while loading, its not raising >> this when i run the script directly fromt the shell. >> >> Can you please help me in this regard. >> >> >> Regards >> >> Abbasi >> >> On Fri, Oct 6, 2017 at 8:01 PM, Bilal Abbasi wrote: >> >>> Thanks i will try that way. Highly appreciated your response. >>> >>> Regards >>> Abbasi >>> >>> On Fri, Oct 6, 2017 at 7:20 PM, Yuriy Gorlichenko >>> wrote: >>> >>>> i think in your case will be better to use bg_api command and run >>>> handler there. >>>> Regarding native or not. Here is a good question - what will be faster >>>> and i not sure that native will be faster way than lua binding >>>> >>>> On Oct 6, 2017 17:05, "Bilal Abbasi" wrote: >>>> >>>>> Hi, >>>>> Thanks for the answer, >>>>> 1) Yes i am aware of that, in fact we need to inform the user that >>>>> file is uploaded successfully or not, so it's normal to have few seconds >>>>> delay. >>>>> 2)Yes i tried that way, and i did able to do that using lua cURL >>>>> library, but i am only interested to know that may be the native function >>>>> can provide me this, I mean if somebody has build this module there >>>>> should/must be a way to get the status. >>>>> >>>>> Regards >>>>> Abbasi >>>>> >>>>> On Fri, Oct 6, 2017 at 12:55 PM, Yuriy Gorlichenko < >>>>> ovoshlook at gmail.com> wrote: >>>>> >>>>>> First of all - be carefull with file uploading from the dialplan, >>>>>> because it can block call handling while you did not get response. >>>>>> >>>>>> Secondly - you also can use some external lua httpclient binding via >>>>>> require and use it to get response statuses that you need. >>>>>> >>>>>> On Oct 6, 2017 10:00, "Bilal Abbasi" wrote: >>>>>> >>>>>>> Anybody here that can please help me out? >>>>>>> >>>>>>> Regards >>>>>>> Abbasi >>>>>>> >>>>>>> On Wed, Oct 4, 2017 at 3:38 PM, Bilal Abbasi >>>>>>> wrote: >>>>>>> >>>>>>>> Hi Users, >>>>>>>> I am using lua script to actually call the curl_sendfile, i am able >>>>>>>> to successfully POST the file on URL, but i am only curious to know the >>>>>>>> response variables to get the status. >>>>>>>> Like i used curl previously and there are two variables that are >>>>>>>> auto set curl_response_data and curl_response_code, i am looking same in >>>>>>>> the curl_sendfile. >>>>>>>> CAn anybody help me in this? >>>>>>>> >>>>>>>> P.S: i did tried to send file using curl, but i am not aware that >>>>>>>> how to do that using the curl -F(--form) option, i can upload a file using >>>>>>>> commandline linux curl command , but could not mapp the option -F in the >>>>>>>> freeswitch. >>>>>>>> >>>>>>>> Regards >>>>>>>> Abbasi >>>>>>>> >>>>>>> >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 vikas452 at gmail.com Wed Oct 11 10:26:45 2017 From: vikas452 at gmail.com (vikas sharma) Date: Wed, 11 Oct 2017 15:56:45 +0530 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode In-Reply-To: References: Message-ID: I have also found that Freeswitch starts sending RTP on the private ip/port first.Ideally it shall not be there.Please let me know if there is any specific setting for RTP nat handling so that Freeswitch send directly on public ip/port only. thanks On Wed, Oct 11, 2017 at 12:03 PM, vikas sharma wrote: > Hi > > Can i get some suggestions to improve the video call quality and reduce > packet loss with Freeswitch server. > > Thanks > > On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma wrote: > >> Anything that can help me in this regard?? >> >> On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma wrote: >> >>> Hi >>> >>> I am making a linphone to linphone call using Freeswitch server in >>> Defualt mode but i am facing huge RTP loss and the video call is getting >>> stuck in between.I have also found that there is a difference between the >>> RTP packets received on A Leg and sent from B-leg and vice versa. >>> >>> Can somebody help me to understand what exactly happens with the RTP >>> packets on server and why there is a difference in the RTP count?? >>> >>> How can we improve the video call quality in moderate network conditions. >>> >>> Thanks >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Wed Oct 11 18:03:16 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Wed, 11 Oct 2017 11:03:16 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Tony, curious if by this you mean it would just be best to set the proxy_read_timeout value in Nginx to whatever means ’never’, and let TCP handle cleaning up disconnects? I’ll certainly admit I’m no expert here, open to suggestions… On Tue, Oct 10, 2017 at 9:40 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The whole idea of a proxy read timeout is a hack. > Working around it forces more hacks. > Tcp has this covered...... > > On Tue, Oct 10, 2017 at 10:43 PM Chad Phillips > wrote: > >> proxy_read_timeout would be an approach to solve the issue from the Nginx >> config side. I don’t like that solution as much, though, because it >> requires guessing how long the socket might be idle, whereas the keepalives >> can be reliably sent to always prevent the timeout. >> >> On Tue, Oct 10, 2017 at 7:02 PM, Ítalo Rossi >> wrote: >> >>> Are we talking about proxy_read_timeout param? >>> >>> On Tue, Oct 10, 2017 at 8:51 PM, Chad Phillips >>> wrote: >>> >>>> I’ve seen this issue when everyone is audio muted and/or doesn’t >>>> trigger a talking event for more than 60 seconds, which in my use case does >>>> happen. I would agree under normal videoconferencing circumstances it would >>>> be rare to go 60 seconds without a single event being sent along the >>>> websocket. >>>> >>>> On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert >>> > wrote: >>>> >>>>> I am curious, what type of situation causes this problem? I mean, we >>>>> use Verto for a conference, granted without Nginx, and seems like there are >>>>> always packets being sent back and forth even when there is no audio or >>>>> video. >>>>> >>>>> >>>>> >>>>> Robert >>>>> >>>>> >>>>> >>>>> *From:* FreeSWITCH-users [mailto:freeswitch-users- >>>>> bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips >>>>> *Sent:* Friday, October 6, 2017 4:53 PM >>>>> *To:* FreeSWITCH Users Help >>>>> *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto >>>>> websocket >>>>> >>>>> >>>>> >>>>> There are various suggested solutions out there. On the Nginx config >>>>> side, easiest is to increase the proxy_read_timeout setting to >>>>> something high, but that seems like it could result in connections getting >>>>> hung open if the server process goes away. >>>>> >>>>> >>>>> >>>>> Then there’s this module which looks pretty fancy and claims to solve >>>>> the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module >>>>> >>>>> — it requires custom compiling nginx though… >>>>> >>>>> >>>>> >>>>> I think I’ll stick with my current solution, which basically follows >>>>> the other suggestion of having the server send periodic keepalives, except >>>>> that I’m initiating from the client. Steps for others interested: >>>>> >>>>> >>>>> >>>>> 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant >>>>> domains in your user directory >>>>> >>>>> >>>>> >>>>> 2. Do something like this after a successful websocket connection: >>>>> >>>>> >>>>> >>>>> MODULE.keepAliveTimer = setInterval(function() { >>>>> >>>>> verto.rpcClient.call("echo", {keepalive: true}); >>>>> >>>>> }, VERTO_KEEPALIVE_INTERVAL) >>>>> >>>>> >>>>> >>>>> 3. Add some cleanup on websocket close: >>>>> >>>>> >>>>> >>>>> if (MODULE.keepAliveTimer) { >>>>> >>>>> clearInterval(MODULE.keepAliveTimer); >>>>> >>>>> MODULE.keepAliveTimer = null; >>>>> >>>>> } >>>>> >>>>> >>>>> >>>>> Request and response lengths are both under 150 characters, so pretty >>>>> short, and I set the intervals at 50 seconds, since Nginx’s default timeout >>>>> is 60 seconds. >>>>> >>>>> >>>>> >>>>> On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert < >>>>> rmundkowsky at ets.org> wrote: >>>>> >>>>> Have you checked if Nginx has timeout settings and/or keep alive >>>>> settings? >>>>> >>>>> >>>>> >>>>> https://stackoverflow.com/questions/10550558/nginx-tcp- >>>>> websockets-timeout-keepalive-config >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Robert >>>>> >>>>> >>>>> >>>>> *From:* FreeSWITCH-users [mailto:freeswitch-users- >>>>> bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips >>>>> *Sent:* Thursday, October 5, 2017 11:07 PM >>>>> *To:* FreeSWITCH Users Help >>>>> *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket >>>>> >>>>> >>>>> >>>>> I've switched to using Nginx to proxy Verto websockets, and have run >>>>> into a small snag: by default, if Nginx doesn't read any data from a proxy >>>>> backend within 60 seconds, it closes the connection, even for websockets. >>>>> >>>>> >>>>> >>>>> It appears the recommended solution is to have the server send some >>>>> kind of regular keepalive. I poked around in mod_verto.c and found a >>>>> 'request.keepalive' variable, but I'm unclear how to set that in the >>>>> request, and/or if it even accomplishes what I'm wanting. >>>>> >>>>> >>>>> >>>>> I've solved the issue for now by periodically sending a JSON RPC >>>>> 'echo' request along the websocket every 50 seconds, and the reply from the >>>>> server is enough to keep the connection open. This is fine, but I am >>>>> curious if there's a way to do it just from the server side, and if not, if >>>>> it's worth it to add a setting to enable that functionality? >>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> This e-mail and any files transmitted with it may contain privileged >>>>> or confidential information. It is solely for use by the individual for >>>>> whom it is intended, even if addressed incorrectly. If you received this >>>>> e-mail in error, please notify the sender; do not disclose, copy, >>>>> distribute, or take any action in reliance on the contents of this >>>>> information; and delete it from your system. Any other use of this e-mail >>>>> is prohibited. >>>>> >>>>> >>>>> >>>>> Thank you for your compliance. >>>>> ------------------------------ >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> >>>>> http://confluence.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ >>>>> options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------ >>>>> >>>>> This e-mail and any files transmitted with it may contain privileged >>>>> or confidential information. It is solely for use by the individual for >>>>> whom it is intended, even if addressed incorrectly. If you received this >>>>> e-mail in error, please notify the sender; do not disclose, copy, >>>>> distribute, or take any action in reliance on the contents of this >>>>> information; and delete it from your system. Any other use of this e-mail >>>>> is prohibited. >>>>> >>>>> Thank you for your compliance. >>>>> ------------------------------ >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ >>>>> options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ >>>> options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Ítalo Rossi >>> italo at freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Wed Oct 11 18:03:41 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Wed, 11 Oct 2017 11:03:41 -0700 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: Robert, I’m not doing anything fancy. Just the simple situation where everyone is audio muted, or there is complete silence for 60 seconds leads to no data being sent over the websocket, thus the disconnect. On Tue, Oct 10, 2017 at 5:48 PM, Mundkowsky, Robert wrote: > I am surprised given microphones usually pick up low levels of noise and > that Nginx would inspect data packets for data. Are you explicitly stopping > data being sent based on push to talk events or VAD? > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Tuesday, October 10, 2017 7:52 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket > > > > I’ve seen this issue when everyone is audio muted and/or doesn’t trigger a > talking event for more than 60 seconds, which in my use case does happen. I > would agree under normal videoconferencing circumstances it would be rare > to go 60 seconds without a single event being sent along the websocket. > > > > On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert > wrote: > > I am curious, what type of situation causes this problem? I mean, we use > Verto for a conference, granted without Nginx, and seems like there are > always packets being sent back and forth even when there is no audio or > video. > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Friday, October 6, 2017 4:53 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sending keepalives on Verto websocket > > > > There are various suggested solutions out there. On the Nginx config side, > easiest is to increase the proxy_read_timeout setting to something high, > but that seems like it could result in connections getting hung open if the > server process goes away. > > > > Then there’s this module which looks pretty fancy and claims to solve the > problem: https://github.com/yaoweibin/nginx_tcp_proxy_module > > — it requires custom compiling nginx though… > > > > I think I’ll stick with my current solution, which basically follows the > other suggestion of having the server send periodic keepalives, except that > I’m initiating from the client. Steps for others interested: > > > > 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant > domains in your user directory > > > > 2. Do something like this after a successful websocket connection: > > > > MODULE.keepAliveTimer = setInterval(function() { > > verto.rpcClient.call("echo", {keepalive: true}); > > }, VERTO_KEEPALIVE_INTERVAL) > > > > 3. Add some cleanup on websocket close: > > > > if (MODULE.keepAliveTimer) { > > clearInterval(MODULE.keepAliveTimer); > > MODULE.keepAliveTimer = null; > > } > > > > Request and response lengths are both under 150 characters, so pretty > short, and I set the intervals at 50 seconds, since Nginx’s default timeout > is 60 seconds. > > > > On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert > wrote: > > Have you checked if Nginx has timeout settings and/or keep alive settings? > > > > https://stackoverflow.com/questions/10550558/nginx-tcp- > websockets-timeout-keepalive-config > > > > > > > Robert > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Chad Phillips > *Sent:* Thursday, October 5, 2017 11:07 PM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Sending keepalives on Verto websocket > > > > I've switched to using Nginx to proxy Verto websockets, and have run into > a small snag: by default, if Nginx doesn't read any data from a proxy > backend within 60 seconds, it closes the connection, even for websockets. > > > > It appears the recommended solution is to have the server send some kind > of regular keepalive. I poked around in mod_verto.c and found a > 'request.keepalive' variable, but I'm unclear how to set that in the > request, and/or if it even accomplishes what I'm wanting. > > > > I've solved the issue for now by periodically sending a JSON RPC 'echo' > request along the websocket every 50 seconds, and the reply from the server > is enough to keep the connection open. This is fine, but I am curious if > there's a way to do it just from the server side, and if not, if it's worth > it to add a setting to enable that functionality? > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > > > Thank you for your compliance. > ------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > > > Thank you for your compliance. > ------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 sagarmalam at gmail.com Wed Oct 11 13:58:28 2017 From: sagarmalam at gmail.com (sagar malam) Date: Wed, 11 Oct 2017 13:58:28 +0000 Subject: [Freeswitch-users] FreeSWITCH not sending Notify to presence subscribers of Callee Message-ID: Hello, I am working with FS 1.6.19. I am facing a situation when FS is not sending Notify packets to SIP endpoints which have subscribed for callee presence. To explain scenario, I have 3 SIP endpoints : EP 1 :1000 EP2 : 1001 EP3 : 1002 EP3 has subscribed for 1000 and 1002 presence information. When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but not for EP2. Should FS send Notify for EP2 as well ? or this is the functionality of FS itself ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From hartnett.tom at gmail.com Wed Oct 11 16:38:26 2017 From: hartnett.tom at gmail.com (Tom Hartnett) Date: Wed, 11 Oct 2017 12:38:26 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference Message-ID: Greetings all, I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. Can anyone offer any advice? Maybe FS is just not suited to this? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 18:34:11 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 14:34:11 -0400 Subject: [Freeswitch-users] freetdm errors.. In-Reply-To: References: Message-ID: <103ABD7E-F0F8-4B2A-87E7-CA730D9F2A88@jerris.com> you’d have to talk to sangoma about issues with their cards and signaling stack. > On Oct 11, 2017, at 4:09 AM, Babak Yakhchali wrote: > > Hi > I'm using sangoma a200 with freetdm on freeswitch 1.6 stable branch. Sometimes I see errors like this in console: > > 2017-10-11 10:58:32.669207 [ERR] ftmod_sangoma_isdn_stack_out.c:574 [s2c31][2:16] Received frame of 323 bytes, exceeding max size of 300 bytes > > Is this a serious problem? From mike at jerris.com Wed Oct 11 18:36:41 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 14:36:41 -0400 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode In-Reply-To: References: Message-ID: check out the confluence page on handling nat, and make sure your clients are sending the public not the private ip for media addr. > On Oct 11, 2017, at 6:26 AM, vikas sharma wrote: > > I have also found that Freeswitch starts sending RTP on the private ip/port first.Ideally it shall not be there.Please let me know if there is any specific setting for RTP nat handling so that Freeswitch send directly on public ip/port only. > > thanks > > On Wed, Oct 11, 2017 at 12:03 PM, vikas sharma > wrote: > Hi > > Can i get some suggestions to improve the video call quality and reduce packet loss with Freeswitch server. > > Thanks > > On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma > wrote: > Anything that can help me in this regard?? > > On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma > wrote: > Hi > > I am making a linphone to linphone call using Freeswitch server in Defualt mode but i am facing huge RTP loss and the video call is getting stuck in between.I have also found that there is a difference between the RTP packets received on A Leg and sent from B-leg and vice versa. > > Can somebody help me to understand what exactly happens with the RTP packets on server and why there is a difference in the RTP count?? > > How can we improve the video call quality in moderate network conditions. > > Thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 18:37:55 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 14:37:55 -0400 Subject: [Freeswitch-users] Sending keepalives on Verto websocket In-Reply-To: References: Message-ID: nginx has nothing at all to do with the media stream, just the signaling path. > On Oct 10, 2017, at 8:48 PM, Mundkowsky, Robert wrote: > > I am surprised given microphones usually pick up low levels of noise and that Nginx would inspect data packets for data. Are you explicitly stopping data being sent based on push to talk events or VAD?  <> > > Robert > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Chad Phillips > Sent: Tuesday, October 10, 2017 7:52 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Sending keepalives on Verto websocket > > I’ve seen this issue when everyone is audio muted and/or doesn’t trigger a talking event for more than 60 seconds, which in my use case does happen. I would agree under normal videoconferencing circumstances it would be rare to go 60 seconds without a single event being sent along the websocket. > > On Mon, Oct 9, 2017 at 7:38 AM, Mundkowsky, Robert > wrote: > I am curious, what type of situation causes this problem?  I mean, we use Verto for a conference, granted without Nginx, and seems like there are always packets being sent back and forth even when there is no audio or video. <> > > Robert > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Chad Phillips > Sent: Friday, October 6, 2017 4:53 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Sending keepalives on Verto websocket > > There are various suggested solutions out there. On the Nginx config side, easiest is to increase the proxy_read_timeout setting to something high, but that seems like it could result in connections getting hung open if the server process goes away. > > Then there’s this module which looks pretty fancy and claims to solve the problem: https://github.com/yaoweibin/nginx_tcp_proxy_module — it requires custom compiling nginx though… > > I think I’ll stick with my current solution, which basically follows the other suggestion of having the server send periodic keepalives, except that I’m initiating from the client. Steps for others interested: > > 1. Add ‘echo’ to the list of jsonrpc-allowed-methods in any relevant domains in your user directory > > 2. Do something like this after a successful websocket connection: > > MODULE.keepAliveTimer = setInterval(function() { > verto.rpcClient.call("echo", {keepalive: true}); > }, VERTO_KEEPALIVE_INTERVAL) > > 3. Add some cleanup on websocket close: > > if (MODULE.keepAliveTimer) { > clearInterval(MODULE.keepAliveTimer); > MODULE.keepAliveTimer = null; > } > > Request and response lengths are both under 150 characters, so pretty short, and I set the intervals at 50 seconds, since Nginx’s default timeout is 60 seconds. > > On Fri, Oct 6, 2017 at 6:39 AM, Mundkowsky, Robert > wrote: > Have you checked if Nginx has timeout settings and/or keep alive settings? <> > > https://stackoverflow.com/questions/10550558/nginx-tcp-websockets-timeout-keepalive-config > > > Robert > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Chad Phillips > Sent: Thursday, October 5, 2017 11:07 PM > To: FreeSWITCH Users Help > > Subject: [Freeswitch-users] Sending keepalives on Verto websocket > > I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. > > It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. > > I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 18:39:04 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 14:39:04 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: References: Message-ID: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> we currently do not have a media distribution method to other endpoint modules other than mod_conference. You can use one of the stream recording methods to distribute to cdn as long as the delay isn’t an issue. > On Oct 11, 2017, at 12:38 PM, Tom Hartnett wrote: > > Greetings all, > I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. > I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. > Can anyone offer any advice? Maybe FS is just not suited to this? From hartnett.tom at gmail.com Wed Oct 11 18:42:56 2017 From: hartnett.tom at gmail.com (Tom Hartnett) Date: Wed, 11 Oct 2017 14:42:56 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> Message-ID: Thanks for your comment Michael, Would it be reasonable to have a bunch of callers (listeners) spy or eavesdrop on a single call to a dummy endpoint (sender)? On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris wrote: > we currently do not have a media distribution method to other endpoint > modules other than mod_conference. You can use one of the stream recording > methods to distribute to cdn as long as the delay isn’t an issue. > > > On Oct 11, 2017, at 12:38 PM, Tom Hartnett > wrote: > > > > Greetings all, > > I have an application where I could use some expert advice. I'd like to > set up an audio broadcast bridge, where the audio from one call would be > distributed to multiple listeners. Listeners have no send audio. Think a > corporate conference call with only one host. All listener calls are > incoming. > > I could of course set this up with mod_conference and just mute all the > listeners. But this requires a bit of overhead to convert everything to PCM > then back to the listener's codecs individually. I'm working on a > resource-strained embedded system, so I'd like to find a way to simply > bridge the speaker audio to multiple endpoints simultaneously. I have very > tight control over the codecs used, and can make sure they are all the > same. Sort of like live MOH (I'd be using mod_portaudio for the sending > endpoint in fact). Delay must be very low, so converting to a shoutcast > stream and using the MOH function isn't an option either. > > Can anyone offer any advice? Maybe FS is just not suited to this? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 18:56:08 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 14:56:08 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> Message-ID: mod_conference would be at least just as efficient as that, probably more. The feature you are looking for doesn’t exist today. > On Oct 11, 2017, at 2:42 PM, Tom Hartnett wrote: > > Thanks for your comment Michael, > Would it be reasonable to have a bunch of callers (listeners) spy or eavesdrop on a single call to a dummy endpoint (sender)? > > On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris > wrote: > we currently do not have a media distribution method to other endpoint modules other than mod_conference. You can use one of the stream recording methods to distribute to cdn as long as the delay isn’t an issue. > > > On Oct 11, 2017, at 12:38 PM, Tom Hartnett > wrote: > > > > Greetings all, > > I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. > > I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. > > Can anyone offer any advice? Maybe FS is just not suited to this? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hartnett.tom at gmail.com Wed Oct 11 19:36:42 2017 From: hartnett.tom at gmail.com (Tom Hartnett) Date: Wed, 11 Oct 2017 15:36:42 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> Message-ID: Sorry to beat a dead horse, but so you mind if I ask why? The info I could discern (which may or may not be accurate) was that eavesdropping uses a simple RTP copy mechanism, while conferencing would involve converting all audio back and forth to PCM for each user (as well as jitter buffers and other overhead for each participant). Is there something I'm missing? On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris wrote: > mod_conference would be at least just as efficient as that, probably > more. The feature you are looking for doesn’t exist today. > > On Oct 11, 2017, at 2:42 PM, Tom Hartnett wrote: > > Thanks for your comment Michael, > Would it be reasonable to have a bunch of callers (listeners) spy or > eavesdrop on a single call to a dummy endpoint (sender)? > > On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris wrote: > >> we currently do not have a media distribution method to other endpoint >> modules other than mod_conference. You can use one of the stream recording >> methods to distribute to cdn as long as the delay isn’t an issue. >> >> > On Oct 11, 2017, at 12:38 PM, Tom Hartnett >> wrote: >> > >> > Greetings all, >> > I have an application where I could use some expert advice. I'd like to >> set up an audio broadcast bridge, where the audio from one call would be >> distributed to multiple listeners. Listeners have no send audio. Think a >> corporate conference call with only one host. All listener calls are >> incoming. >> > I could of course set this up with mod_conference and just mute all the >> listeners. But this requires a bit of overhead to convert everything to PCM >> then back to the listener's codecs individually. I'm working on a >> resource-strained embedded system, so I'd like to find a way to simply >> bridge the speaker audio to multiple endpoints simultaneously. I have very >> tight control over the codecs used, and can make sure they are all the >> same. Sort of like live MOH (I'd be using mod_portaudio for the sending >> endpoint in fact). Delay must be very low, so converting to a shoutcast >> stream and using the MOH function isn't an option either. >> > Can anyone offer any advice? Maybe FS is just not suited to this? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vikas452 at gmail.com Wed Oct 11 19:33:13 2017 From: vikas452 at gmail.com (vikas sharma) Date: Thu, 12 Oct 2017 01:03:13 +0530 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode In-Reply-To: References: Message-ID: Thanks Michael for responding.I will check that. I also found difference in the RTP packet count received on A Leg port of Freeswitch and sent out from B Leg port.I have taken packet dump on server and checked this in RTP streams stats in wireshark. For example 500 video packets were received on A Leg of Freeswitch and then 70 packets were sent out from B Leg to the private ip/port and 350 were sent to public ip/port but still 80 packets are missing.I found a significant difference for bad network condition. Could you please give me some suggestions for this issue?? On 12-Oct-2017 12:07 AM, "Michael Jerris" wrote: check out the confluence page on handling nat, and make sure your clients are sending the public not the private ip for media addr. On Oct 11, 2017, at 6:26 AM, vikas sharma wrote: I have also found that Freeswitch starts sending RTP on the private ip/port first.Ideally it shall not be there.Please let me know if there is any specific setting for RTP nat handling so that Freeswitch send directly on public ip/port only. thanks On Wed, Oct 11, 2017 at 12:03 PM, vikas sharma wrote: > Hi > > Can i get some suggestions to improve the video call quality and reduce > packet loss with Freeswitch server. > > Thanks > > On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma wrote: > >> Anything that can help me in this regard?? >> >> On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma wrote: >> >>> Hi >>> >>> I am making a linphone to linphone call using Freeswitch server in >>> Defualt mode but i am facing huge RTP loss and the video call is getting >>> stuck in between.I have also found that there is a difference between the >>> RTP packets received on A Leg and sent from B-leg and vice versa. >>> >>> Can somebody help me to understand what exactly happens with the RTP >>> packets on server and why there is a difference in the RTP count?? >>> >>> How can we improve the video call quality in moderate network conditions. >>> >>> Thanks >>> >> >> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 19:46:44 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 15:46:44 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> Message-ID: <4743A97F-C698-46B6-A8A8-54513FB04CD8@jerris.com> eavesdrop is going to transcode. its copying the raw already decoded audio from one channel to the other then re-encoding it. > On Oct 11, 2017, at 3:36 PM, Tom Hartnett wrote: > > Sorry to beat a dead horse, but so you mind if I ask why? The info I could discern (which may or may not be accurate) was that eavesdropping uses a simple RTP copy mechanism, while conferencing would involve converting all audio back and forth to PCM for each user (as well as jitter buffers and other overhead for each participant). Is there something I'm missing? > > On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris > wrote: > mod_conference would be at least just as efficient as that, probably more. The feature you are looking for doesn’t exist today. > >> On Oct 11, 2017, at 2:42 PM, Tom Hartnett > wrote: >> >> Thanks for your comment Michael, >> Would it be reasonable to have a bunch of callers (listeners) spy or eavesdrop on a single call to a dummy endpoint (sender)? >> >> On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris > wrote: >> we currently do not have a media distribution method to other endpoint modules other than mod_conference. You can use one of the stream recording methods to distribute to cdn as long as the delay isn’t an issue. >> >> > On Oct 11, 2017, at 12:38 PM, Tom Hartnett > wrote: >> > >> > Greetings all, >> > I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. >> > I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. >> > Can anyone offer any advice? Maybe FS is just not suited to this? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 19:48:07 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 15:48:07 -0400 Subject: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode In-Reply-To: References: Message-ID: <5FD51BF4-91BA-448F-957B-4EC9CED0837B@jerris.com> not more than what i’ve already given, based on the info i have. Sounds like nat issue and its changing the address its sending to, may be loosing some stuff due to packet loss or no where to send it to until it locks on to the right address. Can’t tell any more than that from the info. > On Oct 11, 2017, at 3:33 PM, vikas sharma wrote: > > Thanks Michael for responding.I will check that. > > I also found difference in the RTP packet count received on A Leg port of Freeswitch and sent out from B Leg port.I have taken packet dump on server and checked this in RTP streams stats in wireshark. > > For example 500 video packets were received on A Leg of Freeswitch and then 70 packets were sent out from B Leg to the private ip/port and 350 were sent to public ip/port but still 80 packets are missing.I found a significant difference for bad network condition. > > Could you please give me some suggestions for this issue?? > > On 12-Oct-2017 12:07 AM, "Michael Jerris" > wrote: > check out the confluence page on handling nat, and make sure your clients are sending the public not the private ip for media addr. > >> On Oct 11, 2017, at 6:26 AM, vikas sharma > wrote: >> >> I have also found that Freeswitch starts sending RTP on the private ip/port first.Ideally it shall not be there.Please let me know if there is any specific setting for RTP nat handling so that Freeswitch send directly on public ip/port only. >> >> thanks >> >> On Wed, Oct 11, 2017 at 12:03 PM, vikas sharma > wrote: >> Hi >> >> Can i get some suggestions to improve the video call quality and reduce packet loss with Freeswitch server. >> >> Thanks >> >> On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma > wrote: >> Anything that can help me in this regard?? >> >> On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma > wrote: >> Hi >> >> I am making a linphone to linphone call using Freeswitch server in Defualt mode but i am facing huge RTP loss and the video call is getting stuck in between.I have also found that there is a difference between the RTP packets received on A Leg and sent from B-leg and vice versa. >> >> Can somebody help me to understand what exactly happens with the RTP packets on server and why there is a difference in the RTP count?? >> >> How can we improve the video call quality in moderate network conditions. >> >> Thanks >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hartnett.tom at gmail.com Wed Oct 11 20:16:31 2017 From: hartnett.tom at gmail.com (Tom Hartnett) Date: Wed, 11 Oct 2017 16:16:31 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: <4743A97F-C698-46B6-A8A8-54513FB04CD8@jerris.com> References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> <4743A97F-C698-46B6-A8A8-54513FB04CD8@jerris.com> Message-ID: Thanks for the clarification Michael. I was working off this old post http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-June/002414.html Which I guess is wrong or outdated. On Wed, Oct 11, 2017 at 3:46 PM, Michael Jerris wrote: > eavesdrop is going to transcode. its copying the raw already decoded > audio from one channel to the other then re-encoding it. > > > On Oct 11, 2017, at 3:36 PM, Tom Hartnett wrote: > > Sorry to beat a dead horse, but so you mind if I ask why? The info I could > discern (which may or may not be accurate) was that eavesdropping uses a > simple RTP copy mechanism, while conferencing would involve converting all > audio back and forth to PCM for each user (as well as jitter buffers and > other overhead for each participant). Is there something I'm missing? > > On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris wrote: > >> mod_conference would be at least just as efficient as that, probably >> more. The feature you are looking for doesn’t exist today. >> >> On Oct 11, 2017, at 2:42 PM, Tom Hartnett wrote: >> >> Thanks for your comment Michael, >> Would it be reasonable to have a bunch of callers (listeners) spy or >> eavesdrop on a single call to a dummy endpoint (sender)? >> >> On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris wrote: >> >>> we currently do not have a media distribution method to other endpoint >>> modules other than mod_conference. You can use one of the stream recording >>> methods to distribute to cdn as long as the delay isn’t an issue. >>> >>> > On Oct 11, 2017, at 12:38 PM, Tom Hartnett >>> wrote: >>> > >>> > Greetings all, >>> > I have an application where I could use some expert advice. I'd like >>> to set up an audio broadcast bridge, where the audio from one call would be >>> distributed to multiple listeners. Listeners have no send audio. Think a >>> corporate conference call with only one host. All listener calls are >>> incoming. >>> > I could of course set this up with mod_conference and just mute all >>> the listeners. But this requires a bit of overhead to convert everything to >>> PCM then back to the listener's codecs individually. I'm working on a >>> resource-strained embedded system, so I'd like to find a way to simply >>> bridge the speaker audio to multiple endpoints simultaneously. I have very >>> tight control over the codecs used, and can make sure they are all the >>> same. Sort of like live MOH (I'd be using mod_portaudio for the sending >>> endpoint in fact). Delay must be very low, so converting to a shoutcast >>> stream and using the MOH function isn't an option either. >>> > Can anyone offer any advice? Maybe FS is just not suited to this? >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Oct 11 20:32:06 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 11 Oct 2017 16:32:06 -0400 Subject: [Freeswitch-users] One to many w/o mod_conference In-Reply-To: References: <4B8E2D8B-8A32-4BE3-940E-47531CF487C1@jerris.com> <4743A97F-C698-46B6-A8A8-54513FB04CD8@jerris.com> Message-ID: its not copying rtp packets, its copying decoded audio. > On Oct 11, 2017, at 4:16 PM, Tom Hartnett wrote: > > Thanks for the clarification Michael. I was working off this old post > http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-June/002414.html > Which I guess is wrong or outdated. > > On Wed, Oct 11, 2017 at 3:46 PM, Michael Jerris > wrote: > eavesdrop is going to transcode. its copying the raw already decoded audio from one channel to the other then re-encoding it. > > >> On Oct 11, 2017, at 3:36 PM, Tom Hartnett > wrote: >> >> Sorry to beat a dead horse, but so you mind if I ask why? The info I could discern (which may or may not be accurate) was that eavesdropping uses a simple RTP copy mechanism, while conferencing would involve converting all audio back and forth to PCM for each user (as well as jitter buffers and other overhead for each participant). Is there something I'm missing? >> >> On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris > wrote: >> mod_conference would be at least just as efficient as that, probably more. The feature you are looking for doesn’t exist today. >> >>> On Oct 11, 2017, at 2:42 PM, Tom Hartnett > wrote: >>> >>> Thanks for your comment Michael, >>> Would it be reasonable to have a bunch of callers (listeners) spy or eavesdrop on a single call to a dummy endpoint (sender)? >>> >>> On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris > wrote: >>> we currently do not have a media distribution method to other endpoint modules other than mod_conference. You can use one of the stream recording methods to distribute to cdn as long as the delay isn’t an issue. >>> >>> > On Oct 11, 2017, at 12:38 PM, Tom Hartnett > wrote: >>> > >>> > Greetings all, >>> > I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. >>> > I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. >>> > Can anyone offer any advice? Maybe FS is just not suited to this? >>> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Wed Oct 11 20:24:51 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 11 Oct 2017 21:24:51 +0100 Subject: [Freeswitch-users] doing an att_xfer from within a uuid_standby session (mod_callcenter) Message-ID: Hi, I am trying to make an att_xfer from api. con.execute('att_xfer','sofia/gateway/provider/%s' % dstNumber, uuid) all is working, except when agent presses feature key 0 to bridge both parties, his uuid_standby call hangs-up too and so he is out of the queue and needs to log in again. Is there any solution to this problem? Thanks in advance. Khalil From mario_fs at mgtech.com Wed Oct 11 21:42:55 2017 From: mario_fs at mgtech.com (Mario G) Date: Wed, 11 Oct 2017 14:42:55 -0700 Subject: [Freeswitch-users] FreeSWITCH not sending Notify to presence subscribers of Callee In-Reply-To: References: Message-ID: <16EEC95C-EDA3-45A9-AFA3-19A8CCE3E969@mgtech.com> Sagar, I am not a SIP expert so sorry if this does not match but I wonder if your issue is related one of my MWI issues: FS-10683 . Mario G > On Oct 11, 2017, at 6:58 AM, sagar malam wrote: > > Hello, > > I am working with FS 1.6.19. > > I am facing a situation when FS is not sending Notify packets to SIP endpoints which have subscribed for callee presence. > > To explain scenario, > > I have 3 SIP endpoints : > EP 1 :1000 > EP2 : 1001 > EP3 : 1002 > > EP3 has subscribed for 1000 and 1002 presence information. > > When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but not for EP2. > > Should FS send Notify for EP2 as well ? or this is the functionality of FS itself ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 sagarmalam at gmail.com Thu Oct 12 07:27:30 2017 From: sagarmalam at gmail.com (sagar malam) Date: Thu, 12 Oct 2017 07:27:30 +0000 Subject: [Freeswitch-users] FreeSWITCH not sending Notify to presence subscribers of Callee In-Reply-To: <16EEC95C-EDA3-45A9-AFA3-19A8CCE3E969@mgtech.com> References: <16EEC95C-EDA3-45A9-AFA3-19A8CCE3E969@mgtech.com> Message-ID: Hello Mario, I really appreciate your reply but it is not related to MWI notifications. On Thu, Oct 12, 2017 at 3:13 AM Mario G wrote: > Sagar, I am not a SIP expert so sorry if this does not match but I wonder > if your issue is related one of my MWI issues: FS-10683 > . > Mario G > > On Oct 11, 2017, at 6:58 AM, sagar malam wrote: > > Hello, > > I am working with FS 1.6.19. > > I am facing a situation when FS is not sending Notify packets to SIP > endpoints which have subscribed for callee presence. > > To explain scenario, > > I have 3 SIP endpoints : > EP 1 :1000 > EP2 : 1001 > EP3 : 1002 > > EP3 has subscribed for 1000 and 1002 presence information. > > When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but not > for EP2. > > Should FS send Notify for EP2 as well ? or this is the functionality of FS > itself ? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 aphoticwalker at gmail.com Thu Oct 12 08:19:24 2017 From: aphoticwalker at gmail.com (Charles Yu) Date: Thu, 12 Oct 2017 16:19:24 +0800 Subject: [Freeswitch-users] Control register behavior via a lua script when REGISTER? Message-ID: Hi, Is there a way to control register behavior via a lua script when receiving sip REGISTER ? Such like: I can reject this REGISTER if it does not match some condition. Trigger a sofia::register_failure event? Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Oct 12 10:29:33 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 12 Oct 2017 10:29:33 +0000 Subject: [Freeswitch-users] Attended transfer on Softphones not updating number Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A6C57@mbx-01.sysconfig.co.uk> Hi, We're experiencing problems with the CLI not updating when performing attended transfer to softphones (Jitsi, Bria, Zoiper). This problem does not occur when performing attended transfer to a physical phone (Polycom, Cisco, Yealink), it also does not occur on other VoIP platforms. The SIP messaging is different when performing an attended transfer to different devices, for softphones such as Jitsi, Bria and Zoiper there is no update message sent once the transfer completes whereas for physical phones there is. We have narrowed this down to: /src/mod/endpoints/mod_sofia/mod_sofia.c There are a set of if\elseif statements between line 1943 and 2025 which provide different update messages depending on the receiving device. Can someone please confirm if this is a known problem? Do we need to add support for the softphones? If so, I will raise a JIRA and do further testing, perhaps we can come up with a patch. Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From mike at jerris.com Thu Oct 12 13:48:16 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 12 Oct 2017 13:48:16 +0000 Subject: [Freeswitch-users] Control register behavior via a lua script when REGISTER? In-Reply-To: References: Message-ID: yes, lua directory hooks. On Thu, Oct 12, 2017 at 4:23 AM Charles Yu wrote: > Hi, > > Is there a way to control register behavior via a lua script when > receiving sip REGISTER ? Such like: I can reject this REGISTER if it does > not match some condition. Trigger a sofia::register_failure event? Any > ideas? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 vma at vallimamod.org Fri Oct 13 09:27:07 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 13 Oct 2017 11:27:07 +0200 Subject: [Freeswitch-users] Choppy and fragmented return-audio problem using Unicast In-Reply-To: <59DD3DB9.3070501@voxter.com> References: <59D7DF2C.3010904@voxter.com> <2D82A1C0-6836-4155-89FF-A22341CF325F@vallimamod.org> <59DD3DB9.3070501@voxter.com> Message-ID: <07EE4717-27BB-47EF-8315-81EC0EC43C4E@vallimamod.org> Hi, This log message can come from other modules too. Can you confirm it is from file switch_ivr.c near line 447? If so, that's definitely strange. What version of freeswitch are you using? Can you post your unicast server and esl test code in a gist? I will try it with my system. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 10 Oct 2017, at 23:38, Dayton Turner wrote: > > Hi! > > I thought that first as well - however we *are* specifying the native flag, and the received audio is definitely PCMU. When I capture it bit-for-bit, and import it into audacity, selecting 8khz mono u-Law makes the audio play back perfectly. > > I DO see the "Raw Codec Activation Success L16@" in the logs, despite the received audio stream being PCMU, and despite specifying the native flag. > > We've also attempted streaming L16 back instead of PCMU and its completely incomprehensible static. Whereas now, you can hear it properly, its just stuttered like mad. So, because of this I believe the codec is correct. > > Dayton > > Vallimamod Abdullah wrote: >> Hi, >> >> I think you are getting a codec mismatch. Here is how the unicast command works according to the source code: >> - If native flag is not set, fs will use the L16 codec to exchange media beetween the session and the unicast socket. You can search for "Raw Codec Activation Success L16@" in the debug log to confirm it. >> - If the native flag is set, fs will use the same read and write codec as your current session: the rtp buffer is sent as-is to the unicast socket and the data received from the unicast socket is also sent back as-is to the session. >> >> Hope this helps! >> >> Best Regards, From ynasida at gmail.com Fri Oct 13 15:07:15 2017 From: ynasida at gmail.com (Yuriy Nasida) Date: Fri, 13 Oct 2017 18:07:15 +0300 Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? Message-ID: Hi, anybody able to open? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jkomar at jbox.ca Fri Oct 13 15:08:46 2017 From: jkomar at jbox.ca (Jason Komar) Date: Fri, 13 Oct 2017 09:08:46 -0600 Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? In-Reply-To: References: Message-ID: I wasn't able to access it this morning either. Jason On Fri, Oct 13, 2017 at 9:07 AM, Yuriy Nasida wrote: > Hi, anybody able to open? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 krice at freeswitch.org Fri Oct 13 15:09:30 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 13 Oct 2017 10:09:30 -0500 Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? In-Reply-To: References: Message-ID: <0b3001d34435$4599d020$d0cd7060$@freeswitch.org> We had a small technical issue this moring. Things are back online or are coming online now. If you want to help prevent things like this from happening again donate to the hardware for freeswitch.org fund via the donate button on FreeSWITCH.org From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yuriy Nasida Sent: Friday, October 13, 2017 10:07 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? Hi, anybody able to open? -------------- next part -------------- An HTML attachment was scrubbed... URL: From v.zakhozhai at gmail.com Fri Oct 13 15:10:37 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Fri, 13 Oct 2017 18:10:37 +0300 Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? In-Reply-To: References: Message-ID: Hi, Unfortunately I cannot access it too :( On Fri, Oct 13, 2017 at 6:08 PM, Jason Komar wrote: > I wasn't able to access it this morning either. > > Jason > > > On Fri, Oct 13, 2017 at 9:07 AM, Yuriy Nasida wrote: > >> Hi, anybody able to open? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- С уважением, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From ms4esl at gmail.com Fri Oct 13 14:14:13 2017 From: ms4esl at gmail.com (Marcin S) Date: Fri, 13 Oct 2017 16:14:13 +0200 Subject: [Freeswitch-users] record_session without 3rd party from eavesdrop Message-ID: Hello, I have an C ESL application, that makes calls to two parties (bgapi originate ...), then connects them together (uuid_bridge) and record (record_session). This works fine, but sometimes there is an 3rd leg attached to this bridge via eavesdrop or threeway. The problem is, that when eavesdrop is switched to WHISPER or THREEWAY mode, this 3rd party is also recorded. How can I make recordings without this 3rd party? I have also tried making two separate record_session calls for each of two legs (with RECORD_READ_ONLY), but it doesn't help. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Fri Oct 13 15:20:26 2017 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Fri, 13 Oct 2017 15:20:26 +0000 Subject: [Freeswitch-users] http://files.freeswitch.org/ down ? In-Reply-To: References: Message-ID: <8055367D-1D37-402B-8741-394C0749F107@gmx.net> Checked an hour ago, didn't connect. Now it does. European ISP. Am 13. Oktober 2017 17:10:37 MESZ schrieb Vladyslav Zakhozhai : >Hi, > >Unfortunately I cannot access it too :( > >On Fri, Oct 13, 2017 at 6:08 PM, Jason Komar wrote: > >> I wasn't able to access it this morning either. >> >> Jason >> >> >> On Fri, Oct 13, 2017 at 9:07 AM, Yuriy Nasida >wrote: >> >>> Hi, anybody able to open? >>> >>> >_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From khamlichi.khalil at gmail.com Fri Oct 13 10:25:42 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Fri, 13 Oct 2017 11:25:42 +0100 Subject: [Freeswitch-users] doing an att_xfer from within a uuid_standby session (mod_callcenter) In-Reply-To: References: Message-ID: Hi again, Ok, let me change my question a little bit, how can I do an attended transfer if I am working with mod_callcenter and agent is in uuid_standby mode ? Hope this question is less specific and will receive more suggestions. Thanks in advance. On Wed, Oct 11, 2017 at 9:24 PM, Khalil Khamlichi wrote: > Hi, > > I am trying to make an att_xfer from api. > > con.execute('att_xfer','sofia/gateway/provider/%s' % dstNumber, uuid) > > all is working, except when agent presses feature key 0 to bridge both > parties, his uuid_standby call hangs-up too and so he is out of the > queue and needs to log in again. > > Is there any solution to this problem? > > Thanks in advance. > > Khalil From shaun.stokes at itec-support.co.uk Fri Oct 13 07:35:41 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 13 Oct 2017 07:35:41 +0000 Subject: [Freeswitch-users] Problems accessing files.freeswitch.org Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A70A0@mbx-01.sysconfig.co.uk> files.freeswitch.org (209.105.235.7) appears to be down and freeswitch.org (209.105.235.6) is running like a slug. Presume there is some ongoing server maintenance? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From mike at jerris.com Fri Oct 13 20:29:32 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 13 Oct 2017 16:29:32 -0400 Subject: [Freeswitch-users] Problems accessing files.freeswitch.org In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A70A0@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A70A0@mbx-01.sysconfig.co.uk> Message-ID: <8A67188C-5052-4AB6-92AD-76118423A336@jerris.com> there was an issue much earlier today in the data center, but thats long resolved. Not seeing any issues now on this side. > On Oct 13, 2017, at 3:35 AM, Shaun Stokes wrote: > > files.freeswitch.org (209.105.235.7) appears to be down and freeswitch.org (209.105.235.6) is running like a slug. > > Presume there is some ongoing server maintenance? > From sagarmalam at gmail.com Sun Oct 15 06:32:00 2017 From: sagarmalam at gmail.com (sagar malam) Date: Sun, 15 Oct 2017 06:32:00 +0000 Subject: [Freeswitch-users] FreeSWITCH not sending Notify to presence subscribers of Callee In-Reply-To: References: <16EEC95C-EDA3-45A9-AFA3-19A8CCE3E969@mgtech.com> Message-ID: Hello , Can anyone just point out if this can be an issue with configuration or it is a bug in FS itself ? Thanks in advance On Thu, Oct 12, 2017 at 12:57 PM sagar malam wrote: > Hello Mario, > > I really appreciate your reply but it is not related to MWI notifications. > > On Thu, Oct 12, 2017 at 3:13 AM Mario G wrote: > >> Sagar, I am not a SIP expert so sorry if this does not match but I wonder >> if your issue is related one of my MWI issues: FS-10683 >> . >> Mario G >> >> On Oct 11, 2017, at 6:58 AM, sagar malam wrote: >> >> Hello, >> >> I am working with FS 1.6.19. >> >> I am facing a situation when FS is not sending Notify packets to SIP >> endpoints which have subscribed for callee presence. >> >> To explain scenario, >> >> I have 3 SIP endpoints : >> EP 1 :1000 >> EP2 : 1001 >> EP3 : 1002 >> >> EP3 has subscribed for 1000 and 1002 presence information. >> >> When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but not >> for EP2. >> >> Should FS send Notify for EP2 as well ? or this is the functionality of >> FS itself ? >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/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 mike at jerris.com Sun Oct 15 07:16:59 2017 From: mike at jerris.com (Michael Jerris) Date: Sun, 15 Oct 2017 07:16:59 +0000 Subject: [Freeswitch-users] FreeSWITCH not sending Notify to presence subscribers of Callee In-Reply-To: References: <16EEC95C-EDA3-45A9-AFA3-19A8CCE3E969@mgtech.com> Message-ID: i suspect if this was generally happening i would have bugs filed on it and a lot more complaints On Sun, Oct 15, 2017 at 2:36 AM sagar malam wrote: > Hello , > > Can anyone just point out if this can be an issue with configuration or it > is a bug in FS itself ? > > Thanks in advance > > On Thu, Oct 12, 2017 at 12:57 PM sagar malam wrote: > >> Hello Mario, >> >> I really appreciate your reply but it is not related to MWI notifications. >> >> On Thu, Oct 12, 2017 at 3:13 AM Mario G wrote: >> >>> Sagar, I am not a SIP expert so sorry if this does not match but I >>> wonder if your issue is related one of my MWI issues: FS-10683 >>> . >>> Mario G >>> >>> On Oct 11, 2017, at 6:58 AM, sagar malam wrote: >>> >>> Hello, >>> >>> I am working with FS 1.6.19. >>> >>> I am facing a situation when FS is not sending Notify packets to SIP >>> endpoints which have subscribed for callee presence. >>> >>> To explain scenario, >>> >>> I have 3 SIP endpoints : >>> EP 1 :1000 >>> EP2 : 1001 >>> EP3 : 1002 >>> >>> EP3 has subscribed for 1000 and 1002 presence information. >>> >>> When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but >>> not for EP2. >>> >>> Should FS send Notify for EP2 as well ? or this is the functionality of >>> FS itself ? >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 dig1234 at gmail.com Sun Oct 15 23:57:11 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Sun, 15 Oct 2017 19:57:11 -0400 Subject: [Freeswitch-users] Temporary Equivalent of fs_path In-Reply-To: References: Message-ID: the value you set needs to be a full sip uri, in contrast to fs_path which takes an IP. see this page: https://wiki.freeswitch.org/wiki/Variable_sip_invite_route_uri On Thu, Oct 5, 2017 at 11:18 AM, Colin Morelli wrote: > I had tried this previously and the call simply didn't connect. Freeswitch > responded with a 503, and there wasn't much in the way of logging that was > helpful to debug the issue. I'm sure I'm missing something obvious here. > I'll see if I can get any more info on it, though. > > Best, > Colin > > > On Mon, Oct 2, 2017 at 4:43 PM, Daniel Greenwald > wrote: > >> I think you are looking for: sip_invite_route_uri >> >> On Mon, Sep 25, 2017 at 11:10 AM, Colin Morelli >> wrote: >> >>> Hey all, >>> >>> Trying to figure out how to get the equivalent behavior of fs_path, but >>> only for a single transaction. In other words, I want to start a SIP >>> request hitting a particular proxy, and then simply let Record-Route >>> headers determine where the call should be routed after that. >>> >>> I might be completely overthinking this, but everything I've tried so >>> far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use >>> the given route for all subsequent requests, rather than simply falling >>> back to the session route. >>> >>> Am I missing something? >>> >>> Best, >>> Colin >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 godson.g at gmail.com Mon Oct 16 06:45:08 2017 From: godson.g at gmail.com (Godson Gera) Date: Mon, 16 Oct 2017 12:15:08 +0530 Subject: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on windows In-Reply-To: References: Message-ID: Hi Bipin, I ran into some thing like this once. Turned out to be a the webservice which is called by xml_curl took too long to repond and FS waited there forever with out timing out. Later on timeout option is introduced in xml curl config. Try setting a reasonable timeout and see if that resolves your issue. On Fri, Oct 6, 2017 at 7:39 PM, Bipin Patel wrote: > hi, > > i have two instances of FS running on a windows server, one is used for > routing to carriers and that uses xml files and that works fine, the second > instance is set up to allow clients to register to it using tls and srtp > and users are authenticated using xml_curl which calls a php script which > inturn sends the directory users details on a register from a client. The > problem is every few hours or so FS stops accepting new clients unless i > restart the service, i check the php script and the webserver and those are > running just fine so no idea whats causing FS to stop calling the script or > something else. > > > -- > Regards, > Bipin > > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Thanks & Regards, Godson Gera FreeSWITCH Consultant -------------- next part -------------- An HTML attachment was scrubbed... URL: From cong.wang.itsherpa at gmail.com Mon Oct 16 01:32:14 2017 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Mon, 16 Oct 2017 10:32:14 +0900 Subject: [Freeswitch-users] Connection problem between FreeSwitch Server and Asterisk Server Message-ID: <123BA37E-B431-4321-BC81-9CE04FA0FA2B@gmail.com> I had built a set of server which includes a FS server and an Asterisk Server, and I hope to use FS on application call, Asterisk on public call. Both 2 servers had installed 2 Ethernet controllers(one for public network, and one for LAN). During test I found the App-App call works, but App-public not. Asterisk and FreeSwitch can’t communicate well via LAN. I thought it might a firewall problem, but it still not work since I disabled the iptables on both server. FS is listening a public ip with 5080 and a local ip with 5060, while Asterisk sends the message by local ip with 5080. In this situation, FS can’t receive messages from Asterisk. After I set port to 5060 on Asterisk, it can’t receive messages from FS. Anyone has idea about settings? Do I need to make another sip profile to listen local:5080? From supun.12 at cse.mrt.ac.lk Sat Oct 14 07:57:25 2017 From: supun.12 at cse.mrt.ac.lk (Supun Madushanka) Date: Sat, 14 Oct 2017 13:27:25 +0530 Subject: [Freeswitch-users] OutGoing Sip Call Message-ID: Hi All I am setup freeswitch in aws instance. i can get locally call successfully. now i had an issue i have a sip address "user at sip.linphone.org" i want to get call to that address from locally "1003 at ip". can you help me to config freeswitch for achieve that call Thank you -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12 at cse.mrt.ac.lk -------------- next part -------------- An HTML attachment was scrubbed... URL: From Shawn.Wheeler at interlockconcepts.com Tue Oct 17 00:24:38 2017 From: Shawn.Wheeler at interlockconcepts.com (Shawn Wheeler) Date: Tue, 17 Oct 2017 00:24:38 +0000 Subject: [Freeswitch-users] newbie Question in two parts Message-ID: I have three separate FS systems I running. I would like to be able to make a Mad Boss call from server A to Servers B and C. I found this page that tells how to connect two boxes. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes Question 1 - on this page the two server have difference extension. Is there a way to do this if they similar e.g. 1001, 1002 etc.. Question 2 - Can I make a Mad Boss call from one to multiple servers? If yes, can you point me in a direction to learn how to accomplish this task? Thank you in advance. Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Oct 17 06:54:11 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 17 Oct 2017 10:54:11 +0400 Subject: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on windows In-Reply-To: References: Message-ID: hi, the webserver and server where FS runs is same, for now i have added the timeout options in xml_curl so lets see if it happens again and ill report back, the other issue is for no reason at times FS just hangs or so, running the fs_cli app shows blank, this happened today morning itself, after killing the process and restarting the service it went back to normal Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] FS weird issue with TLS/SRTP and xml_curl on windows From: Godson Gera To: FreeSWITCH Users Help Date: 10/16/2017, 10:45:08 AM > Hi Bipin, > > I ran into some thing like this once. Turned out to be a the > webservice which is called by xml_curl took too long to repond and FS > waited there forever with out timing out. Later on timeout option is > introduced in xml curl config. Try setting a reasonable timeout and > see if that resolves your issue. > > On Fri, Oct 6, 2017 at 7:39 PM, Bipin Patel > wrote: > > hi, > > i have two instances of FS running on a windows server, one is > used for routing to carriers and that uses xml files and that > works fine, the second instance is set up to allow clients to > register to it using tls and srtp and users are authenticated > using xml_curl which calls a php script which inturn sends the > directory users details on a register from a client. The problem > is every few hours or so FS stops accepting new clients unless i > restart the service, i check the php script and the webserver and > those are running just fine so no idea whats causing FS to stop > calling the script or something else. > > > -- > Regards, > Bipin > > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Thanks & Regards, > Godson Gera > FreeSWITCH Consultant > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 josedavid at zennio.com Tue Oct 17 07:12:45 2017 From: josedavid at zennio.com (Jose David Jurado Alonso) Date: Tue, 17 Oct 2017 09:12:45 +0200 Subject: [Freeswitch-users] Calls only between members of the same group In-Reply-To: References: Message-ID: user_context param? 2017-10-09 15:22 GMT+02:00 Ítalo Rossi : > Use different contexts. > > On Mon, Oct 9, 2017 at 9:07 AM, Jose David Jurado Alonso < > josedavid at zennio.com> wrote: > >> Hi, >> >> I would like to restrict all calls so that they can only be made between >> users in the same group (there may be many groups) but I have not found >> anything in particular. >> >> I have been playing with the parameters "toll_allow" and "callgroup" but >> still calling between members of different groups. >> >> Does anyone know how to do this? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew at cassidywebservices.co.uk Tue Oct 17 15:49:01 2017 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 17 Oct 2017 15:49:01 +0000 Subject: [Freeswitch-users] Late Negotiation and ReINVITE Message-ID: Hi Guys, I wonder if this is something anyone else has come across: We make a call out to this particular customer via our wholesale SIP provider. Other end picks up and tries to transfer the call. The call is dropped out end by FreeSWITCH following a reinvite from the remote end with no SDP. FreeSWITCH sends 200 with SDP, other end ACKs then FreeSWITCH drops the call "No answer to offer" We use BT Wholesale's IPEX product and I think we can assume that this call is SIP end-to-end (this is something that I'm finding increasingly common) and what is happening is FreeSWITCH is seeing the reinvite as a late negotiation invite and when the ACK comes back with no SDP, thinks the other end has refused and hanging up the call. I've not tested with inbound-late-negotiation disabled and am hoping to avoid doing so. I have tried setting renegotiate-codec-on-reinvite to false on the outgoing profile which has made no difference. I'm using FreeSWITCH 1.6.19 Kind regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Tue Oct 17 13:34:00 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Tue, 17 Oct 2017 13:34:00 +0000 Subject: [Freeswitch-users] Verto on the iPhone 6 with Safari 11? Message-ID: Just saw the email about "Anthony Minessale Dangerously Demoed Verto on the iPhone 6 with Safari 11 at AstriCon". Glad to use iOS 11 works now. Did you have to do anything unique to get getUserMedia to work? Last time I tried Verto on a iOS 11 beta it did not work. Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From hiastar_alex at 163.com Tue Oct 17 06:59:32 2017 From: hiastar_alex at 163.com (alex) Date: Tue, 17 Oct 2017 14:59:32 +0800 (CST) Subject: [Freeswitch-users] outbound route question Message-ID: <4b75e8fc.66e5.15f29215822.Coremail.hiastar_alex@163.com> Hello: I want to make a loop test from port 1 to port 2. Now, the pri show OK and UP. But the dialplan does not work. the call flow is : SIP A->Port 1->Port 2-> SIP B. it looks SIP A directly call to SIP B without going E1 ports. The attached file are inbound route file 、outbound route file and log file. Is there a problem with outbound route? what should i do? I hope you can help me,thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 00_inbound_did.xml Type: text/xml Size: 696 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: default.xml Type: text/xml Size: 35682 bytes Desc: not available URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: log.exe.txt URL: From mike at jerris.com Tue Oct 17 16:44:05 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 17 Oct 2017 12:44:05 -0400 Subject: [Freeswitch-users] Verto on the iPhone 6 with Safari 11? In-Reply-To: References: Message-ID: <91F07948-615D-4206-88F7-9AD7441F6A0C@jerris.com> Its not in released code yet as there are still quite a number of issues on the safari side making it not production usable. This is still in development. Yes it for sure required changes on the JS side, and still not all functions work properly > On Oct 17, 2017, at 9:34 AM, Mundkowsky, Robert wrote: > > Just saw the email about “Anthony Minessale Dangerously Demoed Verto on the iPhone 6 with Safari 11 at AstriCon”. > > Glad to use iOS 11 works now. Did you have to do anything unique to get getUserMedia to work? > > Last time I tried Verto on a iOS 11 beta it did not work. > > Robert > > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 hartnett.tom at gmail.com Tue Oct 17 17:25:14 2017 From: hartnett.tom at gmail.com (Tom Hartnett) Date: Tue, 17 Oct 2017 13:25:14 -0400 Subject: [Freeswitch-users] Verto on the iPhone 6 with Safari 11? In-Reply-To: <91F07948-615D-4206-88F7-9AD7441F6A0C@jerris.com> References: <91F07948-615D-4206-88F7-9AD7441F6A0C@jerris.com> Message-ID: We're also very interested in getting Verto working with Safari on iOS11. In case it's of any interest we published a voice-based app on iOS that uses Verto. It's a little custom to our product (caller interface for the broadcast industry), but it's free and it *should* allow connection to any Verto signaling server and set up an Opus audio call using the native WebRTC library. https://itunes.apple.com/us/app/comrex-opal-connect/id1276336860?mt=8 On Tue, Oct 17, 2017 at 12:44 PM, Michael Jerris wrote: > Its not in released code yet as there are still quite a number of issues > on the safari side making it not production usable. This is still in > development. Yes it for sure required changes on the JS side, and still > not all functions work properly > > > On Oct 17, 2017, at 9:34 AM, Mundkowsky, Robert > wrote: > > Just saw the email about “Anthony Minessale Dangerously Demoed Verto on > the iPhone 6 with Safari 11 at AstriCon”. > > Glad to use iOS 11 works now. Did you have to do anything unique to get > getUserMedia to work? > > Last time I tried Verto on a iOS 11 beta it did not work. > > Robert > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 asilva at wirelessmundi.com Wed Oct 18 09:47:35 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Wed, 18 Oct 2017 11:47:35 +0200 Subject: [Freeswitch-users] Problems accessing files.freeswitch.org In-Reply-To: <8A67188C-5052-4AB6-92AD-76118423A336@jerris.com> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A70A0@mbx-01.sysconfig.co.uk> <8A67188C-5052-4AB6-92AD-76118423A336@jerris.com> Message-ID: Hi, since the other day issues in files.freeswitch.org, fishye as been down.. https://freeswitch.org/fisheye => 503 Service Temporarily Unavailable accessing from spain. Saludos / Regards / Cumprimentos, António silva On 10/13/2017 10:29 PM, Michael Jerris wrote: > there was an issue much earlier today in the data center, but thats long resolved. Not seeing any issues now on this side. > > >> On Oct 13, 2017, at 3:35 AM, Shaun Stokes wrote: >> >> files.freeswitch.org (209.105.235.7) appears to be down and freeswitch.org (209.105.235.6) is running like a slug. >> >> Presume there is some ongoing server maintenance? >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bilaln018 at gmail.com Wed Oct 18 11:49:40 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 18 Oct 2017 16:49:40 +0500 Subject: [Freeswitch-users] [mod_httpcache][Files Life] Message-ID: Hi Users, I am using mod_http cache to cache my sound files, i can see that once the files are downloaded by the module that remains on the cache folder, is there any way to actually to remove/delete the files after default-max-age. If there is no way in Freeswitch, then i guess i need to write a cronjob to delete the older files, does not seems to have good way to do. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexanderhenryperkins at gmail.com Wed Oct 18 13:54:13 2017 From: alexanderhenryperkins at gmail.com (Alexander Perkins) Date: Wed, 18 Oct 2017 08:54:13 -0500 Subject: [Freeswitch-users] Help with Freeswitch Installation Message-ID: Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From hiastar_alex at 163.com Wed Oct 18 03:45:02 2017 From: hiastar_alex at 163.com (alex) Date: Wed, 18 Oct 2017 11:45:02 +0800 (CST) Subject: [Freeswitch-users] outbound route question Message-ID: <3e21ec25.5359.15f2d95a327.Coremail.hiastar_alex@163.com> I create two sip extension 1000 and 1001,i have to calls 1000 to 1001. This is my outbount route: But it didn't take the route. Why is that? what should i do? This are CLI log: 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1001 at default] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001 at 192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001 at 192.168.1.69:5060! 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at 192.168.1.70! 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001 at 192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001 at 192.168.1.69:5060) Ended 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001 at 192.168.1.69:5060 [CS_DESTROY] 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] Thanks for answer. -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakumarsoftware at gmail.com Wed Oct 18 22:08:43 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Wed, 18 Oct 2017 16:08:43 -0600 Subject: [Freeswitch-users] FS_PATH for Gateways Message-ID: Can we add an fs_path when dialing out via a gateway ? I know this has been asked before and in the archives it has been said to support and also not support fs_path in a gateway. Just wanted to confirm. thank you, Shaks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Oct 19 07:18:30 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 19 Oct 2017 07:18:30 +0000 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: References: Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A9848@mbx-01.sysconfig.co.uk> Is there any particular reason you're using CentOS instead of Debian 8? Debian 8 is the recommended OS for FreeSWITCH for stability and support, the FS Devs use Debian 8 as does a large majority of the FS community. https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Alexander Perkins [alexanderhenryperkins at gmail.com] Sent: 18 October 2017 14:54 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Help with Freeswitch Installation Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From s.safarov at gmail.com Thu Oct 19 07:36:08 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 19 Oct 2017 07:36:08 +0000 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A9848@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A9848@mbx-01.sysconfig.co.uk> Message-ID: Error related to removed repo files from web server. Same error may be happen with Debian dist too. чт, 19 окт. 2017 г. в 10:21, Shaun Stokes : > Is there any particular reason you're using CentOS instead of Debian 8? > > Debian 8 is the recommended OS for FreeSWITCH for stability and support, > the FS Devs use Debian 8 as does a large majority of the FS community. > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie > > ________________________________________ > From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on > behalf of Alexander Perkins [alexanderhenryperkins at gmail.com] > Sent: 18 October 2017 14:54 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Help with Freeswitch Installation > > Hi All. I am attempting to install FS and I am running into issues. I am > attempting to install FS following the instructions on this page- > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. > When I attempt to run the second line, yum install -y > freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am > receiving the following error- http://drops.tiltx.org/BUUzG0. > > When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the > baseURL pointing to > http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when > I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' > error- http://drops.tiltx.org/aqwKWN. > > I have attempted to update the baseURL to > http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as > http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a > 'libpcre.so.1 not found' error. > > I know something is not right, but I do not know what that is. Can > someone point me in the right direction? > > Thank you, > Alex > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Oct 19 08:19:45 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 19 Oct 2017 08:19:45 +0000 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A9848@mbx-01.sysconfig.co.uk>, Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86A9A71@mbx-01.sysconfig.co.uk> Debian 8 repo files are there and can confirm the Debian 8 package installation is working. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Sergey Safarov [s.safarov at gmail.com] Sent: 19 October 2017 08:36 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help with Freeswitch Installation Error related to removed repo files from web server. Same error may be happen with Debian dist too. чт, 19 окт. 2017 г. в 10:21, Shaun Stokes >: Is there any particular reason you're using CentOS instead of Debian 8? Debian 8 is the recommended OS for FreeSWITCH for stability and support, the FS Devs use Debian 8 as does a large majority of the FS community. https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Alexander Perkins [alexanderhenryperkins at gmail.com] Sent: 18 October 2017 14:54 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Help with Freeswitch Installation Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From hiastar_alex at 163.com Thu Oct 19 02:09:14 2017 From: hiastar_alex at 163.com (alex) Date: Thu, 19 Oct 2017 10:09:14 +0800 (CST) Subject: [Freeswitch-users] route question Message-ID: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> Hello: I want to call 1001 by inputting 1001 through this route. But it didn't through the route. Why is that? What should i do? This are CLI log: 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1000 at default] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahabiba at gmail.com Thu Oct 19 10:58:39 2017 From: ahabiba at gmail.com (Ahmed habiba) Date: Thu, 19 Oct 2017 13:58:39 +0300 Subject: [Freeswitch-users] outbound route question In-Reply-To: References: Message-ID: Hi, looks like that the call is handled by another extension, please be sure to put your new extension before the part that is handling the default extension i.e. 1000 to 1009 i.e. before the below line: Thanks, Ahmed Habiba. > > > > From: alex > Subject: [Freeswitch-users] outbound route question > Date: October 18, 2017 at 6:45:02 AM GMT+3 > To: freeswitch-users at lists.freeswitch.org > > > I create two sip extension 1000 and 1001,i have to calls 1000 to 1001. > This is my outbount route: > > > > > > > > > > But it didn't take the route. > Why is that? what should i do? > This are CLI log: > 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public > 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1001 at default] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features > 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001 at 192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] > 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001 at 192.168.1.69:5060! > 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media > 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at 192.168.1.70! > 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001 at 192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001 at 192.168.1.69:5060) Ended > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001 at 192.168.1.69:5060 [CS_DESTROY] > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] > > Thanks for answer. > > > 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>>   > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Thu Oct 19 13:20:17 2017 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Oct 2017 08:20:17 -0500 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: References: Message-ID: <29a601d348dd$01e36b50$05aa41f0$@freeswitch.org> Let me guess, you are using the Amazon version of Centos? They actually break things there by not having $releasever set properly to the version of centos you are actually using, but set it to just “latest” You can edit the file in your /etc/yum.repos.d to set that variable properly From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alexander Perkins Sent: Wednesday, October 18, 2017 8:54 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Help with Freeswitch Installation Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Oct 19 13:30:01 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 19 Oct 2017 13:30:01 +0000 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: <29a601d348dd$01e36b50$05aa41f0$@freeswitch.org> References: <29a601d348dd$01e36b50$05aa41f0$@freeswitch.org> Message-ID: May be creste symbolic link "latest"? чт, 19 окт. 2017 г., 16:21 Ken Rice : > Let me guess, you are using the Amazon version of Centos? > > > > They actually break things there by not having $releasever set properly to > the version of centos you are actually using, but set it to just “latest” > > > > You can edit the file in your /etc/yum.repos.d to set that variable > properly > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Alexander > Perkins > *Sent:* Wednesday, October 18, 2017 8:54 AM > *To:* freeswitch-users at lists.freeswitch.org > > > *Subject:* [Freeswitch-users] Help with Freeswitch Installation > > > > Hi All. I am attempting to install FS and I am running into issues. I am > attempting to install FS following the instructions on this page- > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. > When I attempt to run the second line, yum install -y > freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am > receiving the following error- http://drops.tiltx.org/BUUzG0. > > > > When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the > baseURL pointing to > http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when > I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' > error- http://drops.tiltx.org/aqwKWN. > > > > I have attempted to update the baseURL to > http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as > http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a > 'libpcre.so.1 not found' error. > > > > I know something is not right, but I do not know what that is. Can > someone point me in the right direction? > > > > Thank you, > > Alex > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Thu Oct 19 13:39:29 2017 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Oct 2017 08:39:29 -0500 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: References: <29a601d348dd$01e36b50$05aa41f0$@freeswitch.org> Message-ID: <2ced01d348df$b0ba69b0$122f3d10$@freeswitch.org> That would cause more trouble than it helps, by setting it to latest, they assume everyone is running the same absolute latest version. This would cause people to install packages built for cent7 on cent6 which is itself an issue. People specifically keep certain versions of centos/debian/etc around because it is a known quanity From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Thursday, October 19, 2017 8:30 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help with Freeswitch Installation May be creste symbolic link "latest"? чт, 19 окт. 2017 г., 16:21 Ken Rice >: Let me guess, you are using the Amazon version of Centos? They actually break things there by not having $releasever set properly to the version of centos you are actually using, but set it to just “latest” You can edit the file in your /etc/yum.repos.d to set that variable properly From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Alexander Perkins Sent: Wednesday, October 18, 2017 8:54 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Help with Freeswitch Installation Hi All. I am attempting to install FS and I am running into issues. I am attempting to install FS following the instructions on this page- https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. When I attempt to run the second line, yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am receiving the following error- http://drops.tiltx.org/BUUzG0. When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the baseURL pointing to http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' error- http://drops.tiltx.org/aqwKWN. I have attempted to update the baseURL to http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a 'libpcre.so.1 not found' error. I know something is not right, but I do not know what that is. Can someone point me in the right direction? Thank you, Alex _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Oct 19 13:51:32 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 19 Oct 2017 13:51:32 +0000 Subject: [Freeswitch-users] Help with Freeswitch Installation In-Reply-To: <2ced01d348df$b0ba69b0$122f3d10$@freeswitch.org> References: <29a601d348dd$01e36b50$05aa41f0$@freeswitch.org> <2ced01d348df$b0ba69b0$122f3d10$@freeswitch.org> Message-ID: I agree чт, 19 окт. 2017 г. в 16:40, Ken Rice : > That would cause more trouble than it helps, by setting it to latest, they > assume everyone is running the same absolute latest version. This would > cause people to install packages built for cent7 on cent6 which is itself > an issue. People specifically keep certain versions of centos/debian/etc > around because it is a known quanity > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sergey > Safarov > *Sent:* Thursday, October 19, 2017 8:30 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help with Freeswitch Installation > > > > May be creste symbolic link "latest"? > > > > чт, 19 окт. 2017 г., 16:21 Ken Rice : > > Let me guess, you are using the Amazon version of Centos? > > > > They actually break things there by not having $releasever set properly to > the version of centos you are actually using, but set it to just “latest” > > > > You can edit the file in your /etc/yum.repos.d to set that variable > properly > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Alexander > Perkins > *Sent:* Wednesday, October 18, 2017 8:54 AM > *To:* freeswitch-users at lists.freeswitch.org > > > *Subject:* [Freeswitch-users] Help with Freeswitch Installation > > > > Hi All. I am attempting to install FS and I am running into issues. I am > attempting to install FS following the instructions on this page- > https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7. > When I attempt to run the second line, yum install -y > freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*, I am > receiving the following error- http://drops.tiltx.org/BUUzG0. > > > > When I look at the freeswitch.repo file in /etc/yum.repos.d, I see the > baseURL pointing to > http://files.freeswitch.org/yum-1.6/$releasever/$basearch. However, when > I visit http://files.freeswitch.org/yum-1.6/latest, I get a 'Not Found' > error- http://drops.tiltx.org/aqwKWN. > > > > I have attempted to update the baseURL to > http://files.freeswitch.org/yum-1.6/7/x86_64/ as well as > http://files.freeswitch.org/yum-1.6/7server/x86_64/, but then receive a > 'libpcre.so.1 not found' error. > > > > I know something is not right, but I do not know what that is. Can > someone point me in the right direction? > > > > Thank you, > > Alex > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 vma at vallimamod.org Thu Oct 19 13:58:20 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 19 Oct 2017 15:58:20 +0200 Subject: [Freeswitch-users] route question In-Reply-To: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> References: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> Message-ID: <374CBC52-DD8F-448C-A2F0-A6580C63FE74@vallimamod.org> Hi, - According to your log, you are calling ext 1000 and not 1001: > 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public - If you want to use the regex backreference $1 in an action, you must add a group match in the condition: expression="^(1001)$". If not, it will be taken literally: in your dialplan, you are trying to call "$1" did. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 19 Oct 2017, at 04:09, alex wrote: > > Hello: > I want to call 1001 by inputting 1001 through this route. > > > > > > But it didn't through the route. > Why is that? > What should i do? > This are CLI log: > 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] > 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public > 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1000 at default] > 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default > 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] > 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] > 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) > 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) > 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) > 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) > 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] > 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) > 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] > 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] > 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] > 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended > 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] From jose.lopes at itcenter.com.pt Thu Oct 19 15:11:15 2017 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Thu, 19 Oct 2017 16:11:15 +0100 Subject: [Freeswitch-users] Share Screen stop work after ~20 seconds (FS 1.6.18/1.6.19/master) Message-ID: Hello, I have a Freeswitch WebRTC SIP connected with a Freeswitch MCU. When i use Video Conference on WebRTC, when I share the Screen, the screen is shared, but after 20 seconds the screen freeze and it stops to be presented. The video call of Screen Share is still active. I have this situation after upgrade to FreeSwitch 1.6.18. On Freeswitch 1.6.17 this situation does not happen. I tried also on Freeswitch 1.6.19 and master and this situation still happens. There is any configuration that I need to use on the new versions related to this scenario, so this will not happen? Additional Notes: * On Freeswitch MCU, I made a change that when it is video from Screen Share, the conference member is with flags : mute and deaf. After 20 seconds the video freeze , but after some seconds the screen share start working again. * I tried a Freeswitch WebRTC with MCU, and this situation doesn't happen. Thanks in advance Best regards, *José Lopes* -------------- next part -------------- An HTML attachment was scrubbed... URL: From thetsinling at outlook.com Thu Oct 19 15:11:36 2017 From: thetsinling at outlook.com (bob. chen) Date: Thu, 19 Oct 2017 15:11:36 +0000 Subject: [Freeswitch-users] route question In-Reply-To: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> References: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> Message-ID: CLI log :caller 1000->callee 1000 , open public.xml file, add your extension to it reloadxml try ua dial 1001 agin. 发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 alex 发送时间: 2017年10月19日 10:09 收件人: freeswitch-users at lists.freeswitch.org 主题: [Freeswitch-users] route question Hello: I want to call 1001 by inputting 1001 through this route. But it didn't through the route. Why is that? What should i do? This are CLI log: 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1000 at default] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Thu Oct 19 21:11:00 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 19 Oct 2017 23:11:00 +0200 Subject: [Freeswitch-users] Verto on the iPhone 6 with Safari 11? In-Reply-To: <91F07948-615D-4206-88F7-9AD7441F6A0C@jerris.com> References: <91F07948-615D-4206-88F7-9AD7441F6A0C@jerris.com> Message-ID: Any chance we can test that code? Sent from my iPhone > On 17 Oct 2017, at 18:44, Michael Jerris wrote: > > Its not in released code yet as there are still quite a number of issues on the safari side making it not production usable. This is still in development. Yes it for sure required changes on the JS side, and still not all functions work properly > > >> On Oct 17, 2017, at 9:34 AM, Mundkowsky, Robert wrote: >> >> Just saw the email about “Anthony Minessale Dangerously Demoed Verto on the iPhone 6 with Safari 11 at AstriCon”. >> >> Glad to use iOS 11 works now. Did you have to do anything unique to get getUserMedia to work? >> >> Last time I tried Verto on a iOS 11 beta it did not work. >> >> Robert >> >> >> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >> >> >> Thank you for your compliance. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 anthony.minessale at gmail.com Thu Oct 19 21:52:15 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Oct 2017 16:52:15 -0500 Subject: [Freeswitch-users] Share Screen stop work after ~20 seconds (FS 1.6.18/1.6.19/master) In-Reply-To: References: Message-ID: Please report issues to jira not the mailing list. On Thu, Oct 19, 2017 at 10:11 AM, José Lopes wrote: > Hello, > > I have a Freeswitch WebRTC SIP connected with a Freeswitch MCU. > > When i use Video Conference on WebRTC, when I share the Screen, the screen > is shared, but after 20 seconds the screen freeze and it stops to be > presented. > The video call of Screen Share is still active. > > I have this situation after upgrade to FreeSwitch 1.6.18. > On Freeswitch 1.6.17 this situation does not happen. > I tried also on Freeswitch 1.6.19 and master and this situation still > happens. > > There is any configuration that I need to use on the new versions related > to this scenario, so this will not happen? > > > Additional Notes: > * On Freeswitch MCU, I made a change that when it is video from Screen > Share, the conference member is with flags : mute and deaf. After 20 > seconds the video freeze , but after some seconds the screen share start > working again. > * I tried a Freeswitch WebRTC with MCU, and this situation doesn't happen. > > > Thanks in advance > > > > Best regards, > > *José Lopes* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From nico at vthadden.de Thu Oct 19 23:06:10 2017 From: nico at vthadden.de (Nicola von Thadden) Date: Fri, 20 Oct 2017 01:06:10 +0200 Subject: [Freeswitch-users] Gigaset auto answer Message-ID: Hi, I'm trying to get auto answer working on my Gigaset N510 IP DECT system. I made the changes to the DECT config (enabled "Call Manager" for the extension I want to use_ and tried to call it using the default prefix (8xxxx). Unlike the other phones I work with, the Gigaset device requires authentication for that. Since it is a subscriber to the freeswitch system, the system does not have any information to fulfill it: [ERR] sofia_reg.c:2561 Cannot locate any authentication credentials to complete an authentication request for realm '"test"' The INVITE itself looks as I would expect it to look but the phone answers with:  SIP/2.0 401 Unauthorized         Via: SIP/2.0/UDP 46.38.244.168;rport=5060;branch=z9hG4bKK94yXK0rDUcXj         From: "Nico" ;tag=6gtHSvy9p62HH         To: ;tag=2471884501         Call-ID: dee1f01e-2fc3-1236-8f91-52542680391a         CSeq: 113890892 INVITE         Contact:         WWW-Authenticate: Digest realm="test", nonce="d0533fa35f7a53f86112305a818d732", algorithm=MD5, qop="auth"         User-Agent: N510 IP PRO/42.243.00.000.000         Content-Length: 0 Does anyone have an idea how I can either get freeswitch to retry with authentication or how to get the Gigaset device to accept the call without further auth? Thanks Nico From achinthau at gmail.com Fri Oct 20 04:49:48 2017 From: achinthau at gmail.com (Achintha) Date: Fri, 20 Oct 2017 10:19:48 +0530 Subject: [Freeswitch-users] Freeswitch Crashed Ones A week Message-ID: hi all, we configured freeswitch (1.6.18) with postgresql database (freeswitch DB) and following modules mod_xml_curl : Dynamic Dialplans (from rest Service) mod_json_cdr : for CDR (from rest Service) and one module developed by our self (mod_ards) freeswitch crashed ones a week here i have upload a core dump trace on pastbin https://pastebin.freeswitch.org/view/41e8f799 -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Fri Oct 20 10:22:09 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 20 Oct 2017 12:22:09 +0200 Subject: [Freeswitch-users] Freeswitch Crashed Ones A week In-Reply-To: References: Message-ID: <5BA00617-D521-4722-993C-53BA58B4ACFC@vallimamod.org> Hi, It looks like you have a memory bug ("free(): corrupted unsorted chunks") in your psql odbc driver triggered on disconnect: #3 0x00007febbf38d98e in malloc_printerr (action=1, str=0x7febbf47d380 "free(): corrupted unsorted chunks", ptr=) at malloc.c:4996 #4 0x00007febbf38e696 in _int_free (av=, p=, have_lock=0) at malloc.c:3840 #5 0x00007febb920d328 in QR_free_memory () from /usr/lib/x86_64-linux-gnu/odbc/psqlodbcw.so [...] #13 0x00007febc08d0c79 in switch_odbc_handle_disconnect () from /usr/lib/libfreeswitch.so.1 You can look for a new version of the driver which may correct this bug. Also check why freeswitch gets disconnected from the db. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 20 Oct 2017, at 06:49, Achintha wrote: > > hi all, > > we configured freeswitch (1.6.18) with postgresql database (freeswitch DB) and following modules > mod_xml_curl : Dynamic Dialplans (from rest Service) > mod_json_cdr : for CDR (from rest Service) > and one module developed by our self (mod_ards) > > freeswitch crashed ones a week > here i have upload a core dump trace on pastbin > > https://pastebin.freeswitch.org/view/41e8f799 > > -- > Best Regards.. > Achintha Udukumbura > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vma at vallimamod.org Fri Oct 20 14:01:23 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 20 Oct 2017 16:01:23 +0200 Subject: [Freeswitch-users] Gigaset auto answer In-Reply-To: References: Message-ID: <1F7FD689-BB26-4105-8532-159131C5E4D6@vallimamod.org> Hi, You can define reverse-auth-user and reverse-auth-pass params in the user directory for the reverse authentication. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 20 Oct 2017, at 01:06, Nicola von Thadden wrote: > > Hi, > > > I'm trying to get auto answer working on my Gigaset N510 IP DECT system. > > I made the changes to the DECT config (enabled "Call Manager" for the extension I want to use_ and tried to call it using the default prefix (8xxxx). > > Unlike the other phones I work with, the Gigaset device requires authentication for that. Since it is a subscriber to the freeswitch system, the system does not have any information to fulfill it: > > [ERR] sofia_reg.c:2561 Cannot locate any authentication credentials to complete an authentication request for realm '"test"' > > > The INVITE itself looks as I would expect it to look but the phone answers with: > > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 46.38.244.168;rport=5060;branch=z9hG4bKK94yXK0rDUcXj > From: "Nico" ;tag=6gtHSvy9p62HH > To: ;tag=2471884501 > Call-ID: dee1f01e-2fc3-1236-8f91-52542680391a > CSeq: 113890892 INVITE > Contact: > WWW-Authenticate: Digest realm="test", nonce="d0533fa35f7a53f86112305a818d732", algorithm=MD5, qop="auth" > User-Agent: N510 IP PRO/42.243.00.000.000 > Content-Length: 0 > > > Does anyone have an idea how I can either get freeswitch to retry with authentication or how to get the Gigaset device to accept the call without further auth? > > > Thanks > > Nico > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rick at magicmail.mooo.com Fri Oct 20 14:47:27 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Fri, 20 Oct 2017 15:47:27 +0100 Subject: [Freeswitch-users] IVR missing start of audio files Message-ID: <5ABA7FD2-4DA2-4E01-ACC3-A9C5410F0611@magicmail.mooo.com> I consistently have a problem where the first few seconds of IVR prompts is being missed. I tend to fudge it by putting several seconds of silence at the beginning of each wav file, but this gets to be a bit of a pain. How does everyone else handle this? From rick at magicmail.mooo.com Fri Oct 20 14:52:17 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Fri, 20 Oct 2017 15:52:17 +0100 Subject: [Freeswitch-users] V8 IVR code Message-ID: Using the code at https://freeswitch.org/confluence/display/FREESWITCH/Simple+IVR+in+JavaScript I’m trying to get it not to wait if 1, 2 or 3 is pressed, and put the call straight through. But still wait, if it’s another key (as it might be an extension number). I’ve tried testing the digits variable in the on_dtmf function but just can’t get it to play ball. Anyone else done this, or any pointers? -------------- next part -------------- An HTML attachment was scrubbed... URL: From supun.12 at cse.mrt.ac.lk Fri Oct 20 02:51:13 2017 From: supun.12 at cse.mrt.ac.lk (Supun Madushanka) Date: Fri, 20 Oct 2017 08:21:13 +0530 Subject: [Freeswitch-users] Fwd: freeswitch help In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: Supun Madushanka Date: 20 October 2017 at 08:19 Subject: freeswitch help To: krice at freeswith.org Hi all I install freeswitch in aws instance and i got a sip call internally and externally successfully. tryit.jssip.net this site i locally up and i call throught webrtc ws://34.209.247.97:5066 interall extension as well as extenally. then i install fusionpbx my aws instance. after that some .xml file in /etc/freeswitch has change .xml.noload and also now i cannot connect to the ws://34.209.247.97:5066 could you suggest me something -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12 at cse.mrt.ac.lk -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12 at cse.mrt.ac.lk -------------- next part -------------- An HTML attachment was scrubbed... URL: From hiastar_alex at 163.com Fri Oct 20 03:13:18 2017 From: hiastar_alex at 163.com (alex) Date: Fri, 20 Oct 2017 11:13:18 +0800 (CST) Subject: [Freeswitch-users] route question In-Reply-To: References: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> Message-ID: <7af37e5d.5330.15f37c54f80.Coremail.hiastar_alex@163.com> Hello bob: I put my extension into public.xml.there is still a problem. The telephone is unable to communicate properly. This are CLI log: 2017-10-19 22:53:39.205689 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [94e283be-37d5-4255-97a8-117329c38323] 2017-10-19 22:53:39.205689 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-19 22:53:39.205689 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/1000 [d68525b4-56fd-4bb4-b4c3-2eac19185b44] 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[1000] Calling No:[1000] 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-19 22:53:42.945690 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-19 22:53:46.945689 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/1000 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[1000] Calling No:[1000] 2017-10-19 22:53:46.945689 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/1000 [3add9104-f703-493b-b857-a5a350f55acf] 2017-10-19 22:53:46.945689 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/1000 [CS_INIT] [NO_USER_RESPONSE] 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/1000) Ended 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/1000 [CS_DESTROY] 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/1000) Ended 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/1000 [CS_DESTROY] 2017-10-19 22:53:46.945689 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-19 22:53:46.945689 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-19 22:53:46.945689 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-19 22:53:46.965681 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-19 22:53:46.965681 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] Why is that? Thanks for answer. At 2017-10-19 23:11:36, "bob. chen" wrote: CLI log :caller 1000->callee 1000 , open public.xml file, add your extension to it reloadxml try ua dial 1001 agin. 发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 alex 发送时间: 2017年10月19日 10:09 收件人:freeswitch-users at lists.freeswitch.org 主题: [Freeswitch-users] route question Hello: I want to call 1001 by inputting 1001 through this route. But it didn't through the route. Why is that? What should i do? This are CLI log: 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1000 at default] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Sat Oct 21 03:42:11 2017 From: jungleboogie0 at gmail.com (jungle boogie) Date: Fri, 20 Oct 2017 20:42:11 -0700 Subject: [Freeswitch-users] IVR missing start of audio files In-Reply-To: <5ABA7FD2-4DA2-4E01-ACC3-A9C5410F0611@magicmail.mooo.com> References: <5ABA7FD2-4DA2-4E01-ACC3-A9C5410F0611@magicmail.mooo.com> Message-ID: <1ba36e4d-126e-c44d-04e5-11621578f999@gmail.com> Thus said Rick Jarvis on Fri, 20 Oct 2017 15:47:27 +0100 > I consistently have a problem where the first few seconds of IVR prompts is being missed. I tend to fudge it by putting several seconds of silence at the beginning of each wav file, but this gets to be a bit of a pain. How does everyone else handle this? > How many seconds are you missing? It's not unusual for there to be a delay. See here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/lang/en/demo/demo-ivr.xml#24 From thetsinling at outlook.com Sat Oct 21 06:58:18 2017 From: thetsinling at outlook.com (bob. chen) Date: Sat, 21 Oct 2017 06:58:18 +0000 Subject: [Freeswitch-users] route question In-Reply-To: <7af37e5d.5330.15f37c54f80.Coremail.hiastar_alex@163.com> References: <21b440f4.38cb.15f326449b3.Coremail.hiastar_alex@163.com> <7af37e5d.5330.15f37c54f80.Coremail.hiastar_alex@163.com> Message-ID: Maybe FTDM config error. check this warning from sangoma isdn, ftdm1:1 send an incomp msg.open sangoma isdn msg debug when test. [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) i think ftdm dialplan must process incoming call , because your ftdm 1:1 (outgoing) and ftdm 2:1(incoming) in the same voice card . From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of alex Sent: Friday, October 20, 2017 11:13 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] route question Hello bob: I put my extension into public.xml.there is still a problem. The telephone is unable to communicate properly. This are CLI log: 2017-10-19 22:53:39.205689 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [94e283be-37d5-4255-97a8-117329c38323] 2017-10-19 22:53:39.205689 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-19 22:53:39.205689 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/1000 [d68525b4-56fd-4bb4-b4c3-2eac19185b44] 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[1000] Calling No:[1000] 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-19 22:53:39.205689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-19 22:53:42.945690 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-19 22:53:46.945689 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/1000 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-19 22:53:46.945689 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[1000] Calling No:[1000] 2017-10-19 22:53:46.945689 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/1000 [3add9104-f703-493b-b857-a5a350f55acf] 2017-10-19 22:53:46.945689 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/1000 [CS_INIT] [NO_USER_RESPONSE] 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/1000) Ended 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/1000 [CS_DESTROY] 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/1000) Ended 2017-10-19 22:53:46.945689 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/1000 [CS_DESTROY] 2017-10-19 22:53:46.945689 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-19 22:53:46.945689 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-19 22:53:46.945689 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-19 22:53:46.965681 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-19 22:53:46.965681 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] Why is that? Thanks for answer. At 2017-10-19 23:11:36, "bob. chen" > wrote: CLI log :caller 1000->callee 1000 , open public.xml file, add your extension to it reloadxml try ua dial 1001 agin. 发件人: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] 代表 alex 发送时间: 2017年10月19日 10:09 收件人: freeswitch-users at lists.freeswitch.org 主题: [Freeswitch-users] route question Hello: I want to call 1001 by inputting 1001 through this route. But it didn't through the route. Why is that? What should i do? This are CLI log: 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [714d7752-b04e-4141-b36e-6fbd37f451ae] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context public 2017-10-18 21:49:59.913457 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1000 at default] 2017-10-18 21:49:59.913457 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1000 in context default 2017-10-18 21:49:59.913457 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/1:1/$1 [fb3597b2-86e4-4911-b06e-50c584501a36] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:62 [s1c1][1:1] Outgoing call: Called No:[$1] Calling No:[1000] 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_out.c:79 [s1c1][1:1] Sending SETUP (suId:1 suInstId:1 spInstId:0 dchan:1 ces:0) 2017-10-18 21:49:59.913457 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 [s2c1][2:1] Received SETUP (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:03.893455 [WARNING] ftmod_sangoma_isdn_stack_rcv.c:862 [SNGISDN Q931] s2: Protocol: Unknown Event Code(2): Incomp Msg(276) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1) 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/1:1/$1 [CS_CONSUME_MEDIA] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:267 [s2c1][2:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:0 spInstId:2) 2017-10-18 21:50:07.893455 [INFO] ftmod_sangoma_isdn_stack_hndl.c:153 [s2c1][2:1] Incoming call: Called No:[$1] Calling No:[1000] 2017-10-18 21:50:07.893455 [NOTICE] switch_channel.c:1104 New Channel FreeTDM/2:1/$1 [5b166784-ad11-4117-a352-889c42052a3e] 2017-10-18 21:50:07.893455 [NOTICE] mod_freetdm.c:2733 Hangup FreeTDM/2:1/$1 [CS_INIT] [NO_USER_RESPONSE] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 3 (FreeTDM/2:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/2:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 2 (FreeTDM/1:1/$1) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel FreeTDM/1:1/$1 [CS_DESTROY] 2017-10-18 21:50:07.893455 [INFO] mod_dptools.c:3508 Originate Failed. Cause: NO_USER_RESPONSE 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:385 sofia/internal/1000 at 192.168.1.70 has executed the last dialplan instruction, hanging up. 2017-10-18 21:50:07.893455 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [NORMAL_CLEARING] 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended 2017-10-18 21:50:07.893455 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>> 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Sat Oct 21 14:05:53 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Sat, 21 Oct 2017 19:05:53 +0500 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Why not use proxy param in gateway? On 19 Oct 2017 3:11 am, "Sharath Kumar" wrote: > Can we add an fs_path when dialing out via a gateway ? I know this has > been asked before and in the archives it has been said to support and also > not support fs_path in a gateway. Just wanted to confirm. > > thank you, > Shaks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 gregor at infomedia.si Sun Oct 22 10:52:29 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 22 Oct 2017 12:52:29 +0200 Subject: [Freeswitch-users] Proxy mode, CDR and codec Message-ID: ​Hi! On one FS I am using proxy mode media, just to bridge call from point A to point B. It works ok. But have problems with parsing CDR. How can I check (which variable) what codec was proposed from pont A and which codec was negotiated? Is this possible in proxy mode? Best regards, Gregor ​ -------------- next part -------------- An HTML attachment was scrubbed... URL: From tayeb.meftah at gmail.com Sun Oct 22 11:03:35 2017 From: tayeb.meftah at gmail.com (Tayeb Meftah) Date: Sun, 22 Oct 2017 12:03:35 +0100 Subject: [Freeswitch-users] Proxy mode, CDR and codec In-Reply-To: References: Message-ID: Hello Not pocible, fs only pass media without parcing. Envoyé de mon iPad > Le 22 oct. 2017 à 11:52, Gregor Nanger a écrit : > > ​Hi! > > On one FS I am using proxy mode media, just to bridge call from point A to point B. It works ok. > > But have problems with parsing CDR. How can I check (which variable) what codec was proposed from pont A and which codec was negotiated? Is this possible in proxy mode? > > Best regards, Gregor > > ​ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 nico at vthadden.de Sun Oct 22 16:31:39 2017 From: nico at vthadden.de (Nicola von Thadden) Date: Sun, 22 Oct 2017 18:31:39 +0200 Subject: [Freeswitch-users] Gigaset auto answer In-Reply-To: <1F7FD689-BB26-4105-8532-159131C5E4D6@vallimamod.org> References: <1F7FD689-BB26-4105-8532-159131C5E4D6@vallimamod.org> Message-ID: Hi, thanks for that hint. Unfortunately, I seem to miss something in my config. Starting from a vanilla config I set my domain in vars, change the default password and change directory/default/1012.xml to include the two lines in the params section:           That should be it, right? My freeswitch still complaines when I call 81012 from 1000 about missing credentials: 2017-10-22 18:24:42.318794 [ERR] sofia_reg.c:2616 Cannot locate any authentication credentials to complete an authentication request for realm '"mydomain"'  eval $${domain} also shows the right domain. Should I be suspicious about the doubled quotation mark in the error message? Thanks Nico On 10/20/2017 04:01 PM, Vallimamod Abdullah wrote: > Hi, > > You can define reverse-auth-user and reverse-auth-pass params in the user directory for the reverse authentication. > > Best Regards, From aqsyounas at gmail.com Sun Oct 22 19:35:52 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Mon, 23 Oct 2017 00:35:52 +0500 Subject: [Freeswitch-users] Transcoding on audio file with wav extension and encoded as 8 bit ulaw Message-ID: Greetings list, Just want to confirm will trancoding happen while playing a file having format as below. root at debian:/opt/tmp# soxi song.wav Input File : 'song.wav' Channels : 1 Sample Rate : 8000 Precision : 14-bit Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors File Size : 441k Bit Rate : 64.0k Sample Encoding: 8-bit u-law If i get PCMU as only codec in invite. Here .wav is just a container wrapping PCMU audio. Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From tayeb.meftah at gmail.com Sun Oct 22 20:01:00 2017 From: tayeb.meftah at gmail.com (Tayeb Meftah) Date: Sun, 22 Oct 2017 21:01:00 +0100 Subject: [Freeswitch-users] Transcoding on audio file with wav extension and encoded as 8 bit ulaw In-Reply-To: References: Message-ID: <3F1FB75F-7DDA-4428-BEB0-0E7A132BB249@gmail.com> Hey Logicaly there should be no transcodig needed according to my knoledges Thanks. Envoyé de mon iPad > Le 22 oct. 2017 à 20:35, Aqs Younas a écrit : > > Greetings list, > > Just want to confirm will trancoding happen while playing a file having format as below. > > root at debian:/opt/tmp# soxi song.wav > > Input File : 'song.wav' > Channels : 1 > Sample Rate : 8000 > Precision : 14-bit > Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors > File Size : 441k > Bit Rate : 64.0k > Sample Encoding: 8-bit u-law > > > If i get PCMU as only codec in invite. > > Here .wav is just a container wrapping PCMU audio. > > Best Regards, > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Sun Oct 22 20:02:31 2017 From: abaci64 at gmail.com (Abaci B) Date: Sun, 22 Oct 2017 20:02:31 +0000 Subject: [Freeswitch-users] Transcoding on audio file with wav extension and encoded as 8 bit ulaw In-Reply-To: References: Message-ID: FreeSWITCH would covert the file to raw audio and then re-encode it to PCMU if that's what's negotiated for the call, the only way to eliminate that is to use mod_native_file and make the file a raw PCMU file (song.PCMU) On Sun, Oct 22, 2017 at 7:35 PM, Aqs Younas wrote: > Greetings list, > > Just want to confirm will trancoding happen while playing a file having > format as below. > > root at debian:/opt/tmp# soxi song.wav > > Input File : 'song.wav' > Channels : 1 > Sample Rate : 8000 > Precision : 14-bit > Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors > File Size : 441k > Bit Rate : 64.0k > Sample Encoding: 8-bit u-law > > > If i get PCMU as only codec in invite. > > Here .wav is just a container wrapping PCMU audio. > > Best Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From aqsyounas at gmail.com Sun Oct 22 20:10:48 2017 From: aqsyounas at gmail.com (Aqs Younas) Date: Mon, 23 Oct 2017 01:10:48 +0500 Subject: [Freeswitch-users] Transcoding on audio file with wav extension and encoded as 8 bit ulaw In-Reply-To: References: Message-ID: I think there is raw audio in file. I am following this link as reference. https://superuser.com/questions/670561/what-is-the-ffmpeg-command-to-convert-3gp-to-ulaw On 23 October 2017 at 01:02, Abaci B wrote: > FreeSWITCH would covert the file to raw audio and then re-encode it to > PCMU if that's what's negotiated for the call, the only way to eliminate > that is to use mod_native_file > > and make the file a raw PCMU file (song.PCMU) > > On Sun, Oct 22, 2017 at 7:35 PM, Aqs Younas wrote: > >> Greetings list, >> >> Just want to confirm will trancoding happen while playing a file having >> format as below. >> >> root at debian:/opt/tmp# soxi song.wav >> >> Input File : 'song.wav' >> Channels : 1 >> Sample Rate : 8000 >> Precision : 14-bit >> Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors >> File Size : 441k >> Bit Rate : 64.0k >> Sample Encoding: 8-bit u-law >> >> >> If i get PCMU as only codec in invite. >> >> Here .wav is just a container wrapping PCMU audio. >> >> Best Regards, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From abaci64 at gmail.com Sun Oct 22 20:23:56 2017 From: abaci64 at gmail.com (Abaci B) Date: Sun, 22 Oct 2017 20:23:56 +0000 Subject: [Freeswitch-users] Transcoding on audio file with wav extension and encoded as 8 bit ulaw In-Reply-To: References: Message-ID: not sure which command of that page you use but based on the output of soxi you have a ulaw encoded wav file, what you're looking for is a headerless ulaw file with an extension of PCMU On Sun, Oct 22, 2017 at 8:10 PM, Aqs Younas wrote: > I think there is raw audio in file. I am following this link as reference. > > https://superuser.com/questions/670561/what-is-the- > ffmpeg-command-to-convert-3gp-to-ulaw > > On 23 October 2017 at 01:02, Abaci B wrote: > >> FreeSWITCH would covert the file to raw audio and then re-encode it to >> PCMU if that's what's negotiated for the call, the only way to eliminate >> that is to use mod_native_file >> >> and make the file a raw PCMU file (song.PCMU) >> >> On Sun, Oct 22, 2017 at 7:35 PM, Aqs Younas wrote: >> >>> Greetings list, >>> >>> Just want to confirm will trancoding happen while playing a file having >>> format as below. >>> >>> root at debian:/opt/tmp# soxi song.wav >>> >>> Input File : 'song.wav' >>> Channels : 1 >>> Sample Rate : 8000 >>> Precision : 14-bit >>> Duration : 00:00:55.05 = 440436 samples ~ 4129.09 CDDA sectors >>> File Size : 441k >>> Bit Rate : 64.0k >>> Sample Encoding: 8-bit u-law >>> >>> >>> If i get PCMU as only codec in invite. >>> >>> Here .wav is just a container wrapping PCMU audio. >>> >>> Best Regards, >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 freeswitch940 at gmail.com Mon Oct 23 13:39:49 2017 From: freeswitch940 at gmail.com (Freeswitch user) Date: Mon, 23 Oct 2017 13:39:49 +0000 Subject: [Freeswitch-users] Video recording Message-ID: Hello users, Is there any way in freeswitch to record two way video calls. Please suggest!! Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Tue Oct 24 13:16:45 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Tue, 24 Oct 2017 15:16:45 +0200 Subject: [Freeswitch-users] Gigaset auto answer In-Reply-To: References: <1F7FD689-BB26-4105-8532-159131C5E4D6@vallimamod.org> Message-ID: <749EFE28-B29C-4545-8211-5F86F205EC54@vallimamod.org> Hi, It looks like the reverse lookup didn't work as expected. Is the To domain from the 401 reply the same as your $${domain}? You can also define sip_auth_username and sip_auth_password channel variables in the dialplan, they will be used for the reverse lookup. About the double quotation mark, the source code shows that the realm value is not unquoted when retrieved from the Authentication header. It looks like the sofia lib requires it so to compute the hash. Best Regards -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 22 Oct 2017, at 18:31, Nicola von Thadden wrote: > > Hi, > > > thanks for that hint. Unfortunately, I seem to miss something in my config. > > Starting from a vanilla config I set my domain in vars, change the default password and change directory/default/1012.xml to include the two lines in the params section: > > > > > > That should be it, right? > > My freeswitch still complaines when I call 81012 from 1000 about missing credentials: > > 2017-10-22 18:24:42.318794 [ERR] sofia_reg.c:2616 Cannot locate any authentication credentials to complete an authentication request for realm '"mydomain"' > > eval $${domain} also shows the right domain. > > Should I be suspicious about the doubled quotation mark in the error message? > > > Thanks > > Nico > > > > On 10/20/2017 04:01 PM, Vallimamod Abdullah wrote: >> Hi, >> >> You can define reverse-auth-user and reverse-auth-pass params in the user directory for the reverse authentication. >> >> Best Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From arslansaeed at gmail.com Tue Oct 24 01:32:03 2017 From: arslansaeed at gmail.com (Arslan Saeed) Date: Tue, 24 Oct 2017 12:32:03 +1100 Subject: [Freeswitch-users] Freeswitch Question regarding re-use of previously negotiated codec in response to SIP reINVITE w/o SDP Message-ID: Hello All, Seeking comments/suggestions from the community on a particular Freeswitch (FS) behavior observed. We see that if there is a SIP reINVITE message (in-dialogue) from far end switch coming to FS, it responds back with the last negotiated codec in the SDP offer as opposed to forming a brand new SDP offer with all the supported codecs listed. Now according to my understanding, It is recommended (not mandated) that the SIP entity should respond to a SIP reINVITE w/o SDP with all the Codecs it can support and not just respoind with narrowed down list of codecs (as negotiated in previous SDP offer/answer exchange). This allows the far side a way to re-negotiate the codec from scratch mid dialogue if there is a need (for example call getting transferred to new SIP UA) Any configuration option that can influence this behavior? Regards Arslan -------------- next part -------------- An HTML attachment was scrubbed... URL: From cong.wang.itsherpa at gmail.com Tue Oct 24 03:40:16 2017 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Tue, 24 Oct 2017 12:40:16 +0900 Subject: [Freeswitch-users] Audio delayed for about 3sec. Message-ID: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> Hey all, Today I moved my freeswitch server from VM to cloud server, but I found there has an audio delay on both caller and receiver for about 3sec in each call during call test. Is there any idea for this audio delay? Regards. From cong.wang.itsherpa at gmail.com Tue Oct 24 06:53:11 2017 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Tue, 24 Oct 2017 15:53:11 +0900 Subject: [Freeswitch-users] Audio delayed for about 3sec. In-Reply-To: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> Message-ID: <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Hey all, I had dumped rtp packages and analyzed by wireshark, it shows there has a 5sec clock drift, and the codec between caller and receiver is different (PCMA vs SPEEX). Any idea about this clock drift? Regards. > 2017/10/24 12:40、王聡 のメール: > > Hey all, > > Today I moved my freeswitch server from VM to cloud server, but I found there has an audio delay on both caller and receiver for about 3sec in each call during call test. Is there any idea for this audio delay? > > Regards. From kaiduanx at yahoo.ca Tue Oct 24 13:13:19 2017 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Tue, 24 Oct 2017 13:13:19 +0000 (UTC) Subject: [Freeswitch-users] Pass customized SIP header in INVITE from a leg to b leg References: <1332718271.2353182.1508850799160.ref@mail.yahoo.com> Message-ID: <1332718271.2353182.1508850799160@mail.yahoo.com> Hi, With default mode, for bridged call FreeSwitch strips the non-standard SIP header in INVITE from a leg, is there any configuration to let FreeSwitch pass the header as it is to b leg? Many thanks, /Kaiduan -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Oct 24 15:02:20 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 24 Oct 2017 15:02:20 +0000 Subject: [Freeswitch-users] Audio delayed for about 3sec. In-Reply-To: <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> References: <16D901F8-1A94-4FF3-AAC9-BF37E073BEE4@gmail.com> <97EE2A01-D430-421A-90B6-7CFA71DAF34B@gmail.com> Message-ID: Try configure ntp daemon and then restart freeswitch daemon. Is this help? вт, 24 окт. 2017 г., 17:56 王聡 : > Hey all, > > I had dumped rtp packages and analyzed by wireshark, it shows there has a > 5sec clock drift, and the codec between caller and receiver is different > (PCMA vs SPEEX). Any idea about this clock drift? > > Regards. > > > 2017/10/24 12:40、王聡 のメール: > > > > Hey all, > > > > Today I moved my freeswitch server from VM to cloud server, but I found > there has an audio delay on both caller and receiver for about 3sec in each > call during call test. Is there any idea for this audio delay? > > > > Regards. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakumarsoftware at gmail.com Tue Oct 24 17:04:27 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Tue, 24 Oct 2017 11:04:27 -0600 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Well, that is an option but I found out that the SIP Authentication breaks with certain providers while using the Proxy param. - The Request URI in the REG/INV will reflect the Proxy param. - The Proxy-Auth headers has a URI parameter that will now have the Request URI ( Proxy param). Some SIP providers have the Realm/Domain different from the Proxy. So they end up rejecting the request since the domain in the proxy is not in their white list of authenticated hosts. Eg. Proxy - sip10.voip.xyz.com Realm - xyz.com The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy Auth header will also have the same URI. The Providers I know are unwilling to add the Proxy domain sip10.voip.xyz.com into their white-list of authenticated hosts. They require the Proxy-Auth to have xyz.com. I am not sure if I made myself clear ? Anyway, I managed to set the sip_invite_route_uri as the Proxy. That seemed to work even for gateways. Thank you, Sharath On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: > Why not use proxy param in gateway? > > On 19 Oct 2017 3:11 am, "Sharath Kumar" > wrote: > >> Can we add an fs_path when dialing out via a gateway ? I know this has >> been asked before and in the archives it has been said to support and also >> not support fs_path in a gateway. Just wanted to confirm. >> >> thank you, >> Shaks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 luis.daniel.lucio at gmail.com Tue Oct 24 18:52:27 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 24 Oct 2017 14:52:27 -0400 Subject: [Freeswitch-users] recordFile not working when hangup Message-ID: Before submitting anything, I prefer to do a public question. I have been reported in a Lua script that when people hang up, the recording is not saved in the filesystem. However, when you stop talking I am being reported that it works. The issue on FS 1.6.16 and 1.6.19. Has anyone had the same issue? -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: From servtelar at gmail.com Tue Oct 24 19:52:12 2017 From: servtelar at gmail.com (GM phy) Date: Tue, 24 Oct 2017 16:52:12 -0300 Subject: [Freeswitch-users] webrtc with ecdsa Message-ID: Hello, did anybody get webrtc working with a ECDSA signed cert? I have fs 1.6.19 compiled with openssl 1.0.2g and wss.pem file is -----BEGIN EC PARAMETERS----- BggqhkjOPQMBBw== -----END EC PARAMETERS----- -----BEGIN EC PRIVATE KEY----- MHcCAQEEIPNZ4YUGKQ2cySfnEp1RSOSDp9MTQ21xyISpBmBPRN1/oAoGCCqGSM49 AwEHoUQDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m +A4huOY2MtbwXtBdg2kqjYbcrkAk5zwa6Q== -----END EC PRIVATE KEY----- -----BEGIN CERTIFICATE----- MIIGGTCCBQGgAwIBAgIMAPaMiX8YXElRs/bJMA0GCSqGSIb3DQEBCwUAMEwxCzAJ BgNVBAYTAkJFMRkwFwYDVQQKExBHbG9iYWxTaWduIG52LXNhMSIwIAYDVQQDExlB bHBoYVNTTCBDQSAtIFNIQTI1NiAtIEcyMB4XDTE3MTAxNzEzNDY1MVoXDTE4MTAx ODEzNDY1MVowQTEhMB8GA1UECxMYRG9tYWluIENvbnRyb2wgVmFsaWRhdGVkMRww GgYDVQQDDBMqLmxhY29udGluZW50YWwuY29tMFkwEwYHKoZIzj0CAQYIKoZIzj0D AQcDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m+A4h uOY2MtbwXtBdg2kqjYbcrkAk5zwa6aOCA88wggPLMA4GA1UdDwEB/wQEAwIDiDCB iQYIKwYBBQUHAQEEfTB7MEIGCCsGAQUFBzAChjZodHRwOi8vc2VjdXJlMi5hbHBo .... when we try to connect (we are using sipjs and tried from chrome and firefox last versions) we get a handshake failure. any lead or help will be appreciated, thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From colin.morelli at gmail.com Tue Oct 24 20:41:16 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Tue, 24 Oct 2017 16:41:16 -0400 Subject: [Freeswitch-users] Temporary Equivalent of fs_path In-Reply-To: References: Message-ID: Tried this again today, and it seems to not work. It sets the header of the request just fine, but it doesn't change where the request ends up going to. sip_route_uri does change the destination, but seems to stick for the entire session. Is there anything in the middle? Best, Colin On Sun, Oct 15, 2017 at 7:57 PM, Daniel Greenwald wrote: > the value you set needs to be a full sip uri, in contrast to fs_path which > takes an IP. > see this page: > https://wiki.freeswitch.org/wiki/Variable_sip_invite_route_uri > > On Thu, Oct 5, 2017 at 11:18 AM, Colin Morelli > wrote: > >> I had tried this previously and the call simply didn't connect. >> Freeswitch responded with a 503, and there wasn't much in the way of >> logging that was helpful to debug the issue. I'm sure I'm missing something >> obvious here. I'll see if I can get any more info on it, though. >> >> Best, >> Colin >> >> >> On Mon, Oct 2, 2017 at 4:43 PM, Daniel Greenwald >> wrote: >> >>> I think you are looking for: sip_invite_route_uri >>> >>> On Mon, Sep 25, 2017 at 11:10 AM, Colin Morelli >> > wrote: >>> >>>> Hey all, >>>> >>>> Trying to figure out how to get the equivalent behavior of fs_path, but >>>> only for a single transaction. In other words, I want to start a SIP >>>> request hitting a particular proxy, and then simply let Record-Route >>>> headers determine where the call should be routed after that. >>>> >>>> I might be completely overthinking this, but everything I've tried so >>>> far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use >>>> the given route for all subsequent requests, rather than simply falling >>>> back to the session route. >>>> >>>> Am I missing something? >>>> >>>> Best, >>>> Colin >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 khamlichi.khalil at gmail.com Tue Oct 24 21:00:13 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Tue, 24 Oct 2017 22:00:13 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line Message-ID: Hi, is it possible to use mod_callcenter queues without answering the line, that is the customer is held in the queue and hearing ringing (without the call being answered). In fact this is a mandatory requirement when calls have a cost and we don't want to charge customer for waiting time but only for talk time. Thanks in advance. Khalil From servtelar at gmail.com Tue Oct 24 21:15:50 2017 From: servtelar at gmail.com (GM phy) Date: Tue, 24 Oct 2017 16:15:50 -0500 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: Message-ID: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Hi That’s not a mandatory requirement at all, waiting time counts as part of the call. Doing something like that you’ll have a network disconnection due lack of answer, no matter if you send 180 or 183. The timeout will depend on the scenario on each country, but in general terms it will be between one or two minutes. GM > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi wrote: > > Hi, > > is it possible to use mod_callcenter queues without answering the > line, that is the customer is held in the queue and hearing ringing > (without the call being answered). > > In fact this is a mandatory requirement when calls have a cost and we > don't want to charge customer for waiting time but only for talk time. > > Thanks in advance. > > Khalil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From italo at freeswitch.org Tue Oct 24 22:36:19 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 24 Oct 2017 22:36:19 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: Don't do that. :) Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: > Hi > > That’s not a mandatory requirement at all, waiting time counts as part of > the call. > Doing something like that you’ll have a network disconnection due lack of > answer, no matter if you send 180 or 183. The timeout will depend on the > scenario on each country, but in general terms it will be between one or > two minutes. > > GM > > > > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi < > khamlichi.khalil at gmail.com> wrote: > > > > Hi, > > > > is it possible to use mod_callcenter queues without answering the > > line, that is the customer is held in the queue and hearing ringing > > (without the call being answered). > > > > In fact this is a mandatory requirement when calls have a cost and we > > don't want to charge customer for waiting time but only for talk time. > > > > Thanks in advance. > > > > Khalil > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 abaci64 at gmail.com Tue Oct 24 22:58:58 2017 From: abaci64 at gmail.com (Abaci B) Date: Tue, 24 Oct 2017 22:58:58 +0000 Subject: [Freeswitch-users] recordFile not working when hangup In-Reply-To: References: Message-ID: it can what the lua script does after hangup that causes it, so the best way to test would be to a test with a simple lua file that just answers the call and records a file and see for yourself if it's saved after hangup, if possible also monitor the console for any logs. you can do something like session:answer(); session:recordFile('/tmp/test.wav'); On Tue, Oct 24, 2017 at 6:52 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Before submitting anything, I prefer to do a public question. I have been > reported in a Lua script that when people hang up, the recording is not > saved in the filesystem. However, when you stop talking I am being reported > that it works. The issue on FS 1.6.16 and 1.6.19. > > Has anyone had the same issue? > > -- > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Oct 25 05:44:23 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 25 Oct 2017 05:44:23 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: I made this on my server by changing module source code. As say GM phy is importand to get answer from agent durring 1-2 minutes after call queried ср, 25 окт. 2017 г., 1:37 Ítalo Rossi : > Don't do that. :) > Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: > >> Hi >> >> That’s not a mandatory requirement at all, waiting time counts as part of >> the call. >> Doing something like that you’ll have a network disconnection due lack of >> answer, no matter if you send 180 or 183. The timeout will depend on the >> scenario on each country, but in general terms it will be between one or >> two minutes. >> >> GM >> >> >> > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi < >> khamlichi.khalil at gmail.com> wrote: >> > >> > Hi, >> > >> > is it possible to use mod_callcenter queues without answering the >> > line, that is the customer is held in the queue and hearing ringing >> > (without the call being answered). >> > >> > In fact this is a mandatory requirement when calls have a cost and we >> > don't want to charge customer for waiting time but only for talk time. >> > >> > Thanks in advance. >> > >> > Khalil >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 alexandr.popov at iqoption.com Wed Oct 25 06:39:28 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Wed, 25 Oct 2017 09:39:28 +0300 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Why not use realm param in gateway? 2017-10-24 20:04 GMT+03:00 Sharath Kumar : > Well, that is an option but I found out that the SIP Authentication breaks > with certain providers while using the Proxy param. > > - The Request URI in the REG/INV will reflect the Proxy param. > - The Proxy-Auth headers has a URI parameter that will now have the > Request URI ( Proxy param). > > Some SIP providers have the Realm/Domain different from the Proxy. So they > end up rejecting the request since the domain in the proxy is not in their > white list of authenticated hosts. > > Eg. Proxy - sip10.voip.xyz.com > Realm - xyz.com > > The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy > Auth header will also have the same URI. The Providers I know are unwilling > to add the Proxy domain sip10.voip.xyz.com into their white-list of > authenticated hosts. They require the Proxy-Auth to have xyz.com. > > I am not sure if I made myself clear ? > > Anyway, I managed to set the sip_invite_route_uri as the Proxy. That > seemed to work even for gateways. > > Thank you, > Sharath > > > On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: > >> Why not use proxy param in gateway? >> >> On 19 Oct 2017 3:11 am, "Sharath Kumar" >> wrote: >> >>> Can we add an fs_path when dialing out via a gateway ? I know this has >>> been asked before and in the archives it has been said to support and also >>> not support fs_path in a gateway. Just wanted to confirm. >>> >>> thank you, >>> Shaks. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 ksrigo at gmail.com Wed Oct 25 07:20:34 2017 From: ksrigo at gmail.com (Srigo Kanapathipillai) Date: Wed, 25 Oct 2017 09:20:34 +0200 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Hi Sharath Kumar, Use the params: outbound-proxy and proxy. Eg: Outbound-proxy: sip10.voip.xyz.com Proxy: xyz.com Srigo Sent from my iPhone > On 24 Oct 2017, at 19:04, Sharath Kumar wrote: > > Well, that is an option but I found out that the SIP Authentication breaks with certain providers while using the Proxy param. > > - The Request URI in the REG/INV will reflect the Proxy param. > - The Proxy-Auth headers has a URI parameter that will now have the Request URI ( Proxy param). > > Some SIP providers have the Realm/Domain different from the Proxy. So they end up rejecting the request since the domain in the proxy is not in their white list of authenticated hosts. > > Eg. Proxy - sip10.voip.xyz.com > Realm - xyz.com > > The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy Auth header will also have the same URI. The Providers I know are unwilling to add the Proxy domain sip10.voip.xyz.com into their white-list of authenticated hosts. They require the Proxy-Auth to have xyz.com. > > I am not sure if I made myself clear ? > > Anyway, I managed to set the sip_invite_route_uri as the Proxy. That seemed to work even for gateways. > > Thank you, > Sharath > > >> On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: >> Why not use proxy param in gateway? >> >>> On 19 Oct 2017 3:11 am, "Sharath Kumar" wrote: >>> Can we add an fs_path when dialing out via a gateway ? I know this has been asked before and in the archives it has been said to support and also not support fs_path in a gateway. Just wanted to confirm. >>> >>> thank you, >>> Shaks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 Wed Oct 25 07:18:44 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 25 Oct 2017 07:18:44 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> , Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F6EEB3C2@mbx-01.sysconfig.co.uk> You could always create the illusion that the call hasn't been answered yet by using MoH attached to a stream that provides ringing, this would allow you to get around the 1-2 minute call timeout. ________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Sergey Safarov [s.safarov at gmail.com] Sent: 25 October 2017 06:44 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line I made this on my server by changing module source code. As say GM phy is importand to get answer from agent durring 1-2 minutes after call queried ср, 25 окт. 2017 г., 1:37 Ítalo Rossi >: Don't do that. :) Em ter, 24 de out de 2017 às 18:16, GM phy > escreveu: Hi That’s not a mandatory requirement at all, waiting time counts as part of the call. Doing something like that you’ll have a network disconnection due lack of answer, no matter if you send 180 or 183. The timeout will depend on the scenario on each country, but in general terms it will be between one or two minutes. GM > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi > wrote: > > Hi, > > is it possible to use mod_callcenter queues without answering the > line, that is the customer is held in the queue and hearing ringing > (without the call being answered). > > In fact this is a mandatory requirement when calls have a cost and we > don't want to charge customer for waiting time but only for talk time. > > Thanks in advance. > > Khalil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Wed Oct 25 08:32:08 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 25 Oct 2017 09:32:08 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: In our case it is really mandatory, if customer does not get to talk to an agent the company has no legal right to bill him anything. for this we really must implement waiting without answering the call, On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: > Don't do that. :) > > Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >> >> Hi >> >> That’s not a mandatory requirement at all, waiting time counts as part of >> the call. >> Doing something like that you’ll have a network disconnection due lack of >> answer, no matter if you send 180 or 183. The timeout will depend on the >> scenario on each country, but in general terms it will be between one or two >> minutes. >> >> GM >> >> >> > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >> > wrote: >> > >> > Hi, >> > >> > is it possible to use mod_callcenter queues without answering the >> > line, that is the customer is held in the queue and hearing ringing >> > (without the call being answered). >> > >> > In fact this is a mandatory requirement when calls have a cost and we >> > don't want to charge customer for waiting time but only for talk time. >> > >> > Thanks in advance. >> > >> > Khalil >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From khamlichi.khalil at gmail.com Wed Oct 25 08:43:02 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 25 Oct 2017 09:43:02 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: to give you an example, you call to talk to your bank account manager, the bank charges an expensive fee for that, for that the bank is legally not allowed to charge waiting time. On Wed, Oct 25, 2017 at 9:32 AM, Khalil Khamlichi wrote: > In our case it is really mandatory, if customer does not get to talk > to an agent the company has no legal right to bill him anything. for > this we really must implement waiting without answering the call, > > On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >> Don't do that. :) >> >> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>> >>> Hi >>> >>> That’s not a mandatory requirement at all, waiting time counts as part of >>> the call. >>> Doing something like that you’ll have a network disconnection due lack of >>> answer, no matter if you send 180 or 183. The timeout will depend on the >>> scenario on each country, but in general terms it will be between one or two >>> minutes. >>> >>> GM >>> >>> >>> > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>> > wrote: >>> > >>> > Hi, >>> > >>> > is it possible to use mod_callcenter queues without answering the >>> > line, that is the customer is held in the queue and hearing ringing >>> > (without the call being answered). >>> > >>> > In fact this is a mandatory requirement when calls have a cost and we >>> > don't want to charge customer for waiting time but only for talk time. >>> > >>> > Thanks in advance. >>> > >>> > Khalil >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From covici at ccs.covici.com Wed Oct 25 09:22:38 2017 From: covici at ccs.covici.com (John Covici) Date: Wed, 25 Oct 2017 05:22:38 -0400 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: The network will time out in a short period, so no matter what you do it will not work. Just don't start charging until the client is talking to someone, but fs won't help you directly. There may be a way in mod_callcenter to know when an agent picks up and you can update a database when that happens. On Wed, 25 Oct 2017 04:43:02 -0400, Khalil Khamlichi wrote: > > to give you an example, you call to talk to your bank account manager, > the bank charges an expensive fee for that, for that the bank is > legally not allowed to charge waiting time. > > On Wed, Oct 25, 2017 at 9:32 AM, Khalil Khamlichi > wrote: > > In our case it is really mandatory, if customer does not get to talk > > to an agent the company has no legal right to bill him anything. for > > this we really must implement waiting without answering the call, > > > > On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: > >> Don't do that. :) > >> > >> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: > >>> > >>> Hi > >>> > >>> That’s not a mandatory requirement at all, waiting time counts as part of > >>> the call. > >>> Doing something like that you’ll have a network disconnection due lack of > >>> answer, no matter if you send 180 or 183. The timeout will depend on the > >>> scenario on each country, but in general terms it will be between one or two > >>> minutes. > >>> > >>> GM > >>> > >>> > >>> > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi > >>> > wrote: > >>> > > >>> > Hi, > >>> > > >>> > is it possible to use mod_callcenter queues without answering the > >>> > line, that is the customer is held in the queue and hearing ringing > >>> > (without the call being answered). > >>> > > >>> > In fact this is a mandatory requirement when calls have a cost and we > >>> > don't want to charge customer for waiting time but only for talk time. > >>> > > >>> > Thanks in advance. > >>> > > >>> > Khalil > >>> > > >>> > > >>> > _________________________________________________________________________ > >>> > Professional FreeSWITCH Consulting Services: > >>> > consulting at freeswitch.org > >>> > http://www.freeswitchsolutions.com > >>> > > >>> > Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://confluence.freeswitch.org > >>> > http://www.cluecon.com > >>> > > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > >>> > >>> _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://confluence.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From shaun.stokes at itec-support.co.uk Wed Oct 25 09:23:12 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 25 Oct 2017 09:23:12 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> , Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F6EEB473@mbx-01.sysconfig.co.uk> There's a far more simple solution. You will have multiple call legs, the member leg is the inbound leg of the call which is connected to the call centre queue which in your example I would ignore. There will also be an agent leg for each attempt to deliver the call to an agent, for your example I would use the billsec field on the agent leg which will give you the time the agent has spent on the phone with a caller. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Khalil Khamlichi [khamlichi.khalil at gmail.com] Sent: 25 October 2017 09:43 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line to give you an example, you call to talk to your bank account manager, the bank charges an expensive fee for that, for that the bank is legally not allowed to charge waiting time. On Wed, Oct 25, 2017 at 9:32 AM, Khalil Khamlichi wrote: > In our case it is really mandatory, if customer does not get to talk > to an agent the company has no legal right to bill him anything. for > this we really must implement waiting without answering the call, > > On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >> Don't do that. :) >> >> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>> >>> Hi >>> >>> That’s not a mandatory requirement at all, waiting time counts as part of >>> the call. >>> Doing something like that you’ll have a network disconnection due lack of >>> answer, no matter if you send 180 or 183. The timeout will depend on the >>> scenario on each country, but in general terms it will be between one or two >>> minutes. >>> >>> GM >>> >>> >>> > On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>> > wrote: >>> > >>> > Hi, >>> > >>> > is it possible to use mod_callcenter queues without answering the >>> > line, that is the customer is held in the queue and hearing ringing >>> > (without the call being answered). >>> > >>> > In fact this is a mandatory requirement when calls have a cost and we >>> > don't want to charge customer for waiting time but only for talk time. >>> > >>> > Thanks in advance. >>> > >>> > Khalil >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From marko.mrvelj at amphinicy.com Wed Oct 25 09:39:21 2017 From: marko.mrvelj at amphinicy.com (Marko Mrvelj) Date: Wed, 25 Oct 2017 11:39:21 +0200 (CEST) Subject: [Freeswitch-users] Adding authorization to SIP messaging Message-ID: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> Hi all, We are experimenting with chatplan, in order to get basic chat functionality. Messages do work as expected, but there is a problem that message can be submitted to server without any authorization. Could somebody tell me if it is possible to restrict messages to only authorized users? How can this be done? Thanks in advance, Marko From asilva at wirelessmundi.com Wed Oct 25 10:28:18 2017 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 25 Oct 2017 12:28:18 +0200 Subject: [Freeswitch-users] Adding authorization to SIP messaging In-Reply-To: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> References: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> Message-ID: <112898af-eb13-7aec-af47-82e911399bb8@wirelessmundi.com> Hi, yes, you can set in the sofia profile the parameter: On 10/25/2017 11:39 AM, Marko Mrvelj wrote: > Hi all, > > We are experimenting with chatplan, in order to get basic chat > functionality. > > Messages do work as expected, but there is a problem that message can be > submitted to server without any authorization. Could somebody tell me if > it is possible to restrict messages to only authorized users? How can > this be done? > > Thanks in advance, > Marko > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva From rtreleaven at bunnykick.ca Wed Oct 25 13:31:06 2017 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Wed, 25 Oct 2017 09:31:06 -0400 Subject: [Freeswitch-users] Adding authorization to SIP messaging In-Reply-To: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> References: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> Message-ID: On Wed, Oct 25, 2017 at 5:39 AM, Marko Mrvelj wrote: > Hi all, > > We are experimenting with chatplan, in order to get basic chat > functionality. > > Messages do work as expected, but there is a problem that message can be > submitted to server without any authorization. Could somebody tell me if > it is possible to restrict messages to only authorized users? How can > this be done? > > Thanks in advance, > Marko > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Russell Treleaven sip:rtreleaven at sip.bunnykick.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: From servtelar at gmail.com Wed Oct 25 13:40:42 2017 From: servtelar at gmail.com (GM phy) Date: Wed, 25 Oct 2017 08:40:42 -0500 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. > On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: > > In our case it is really mandatory, if customer does not get to talk > to an agent the company has no legal right to bill him anything. for > this we really must implement waiting without answering the call, > > On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >> Don't do that. :) >> >> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>> >>> Hi >>> >>> That’s not a mandatory requirement at all, waiting time counts as part of >>> the call. >>> Doing something like that you’ll have a network disconnection due lack of >>> answer, no matter if you send 180 or 183. The timeout will depend on the >>> scenario on each country, but in general terms it will be between one or two >>> minutes. >>> >>> GM >>> >>> >>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>> wrote: >>>> >>>> Hi, >>>> >>>> is it possible to use mod_callcenter queues without answering the >>>> line, that is the customer is held in the queue and hearing ringing >>>> (without the call being answered). >>>> >>>> In fact this is a mandatory requirement when calls have a cost and we >>>> don't want to charge customer for waiting time but only for talk time. >>>> >>>> Thanks in advance. >>>> >>>> Khalil >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shakumarsoftware at gmail.com Wed Oct 25 17:00:00 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Wed, 25 Oct 2017 11:00:00 -0600 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Alexandr, I am using the realm and realm is correct. But some providers are using the domain Req URI to be part of their white list of domains. So I want the Req URI/URI in Proxy header to reflect "realm". INVITE* sip:bob at biloxi.example.com * SIP/2.0 Proxy-Authorization: Digest username="alice", realm="atlanta.example.com", nonce="wf84f1ceczx41ae6cbe5aea9c8e88d359", opaque="",* uri="sip:bob at biloxi.example.com ",* response="42ce3cef44b22f50c6a6071bc8" On Wed, Oct 25, 2017 at 12:39 AM, Alexandr Popov < alexandr.popov at iqoption.com> wrote: > Why not use realm param in gateway? > > 2017-10-24 20:04 GMT+03:00 Sharath Kumar : > >> Well, that is an option but I found out that the SIP Authentication >> breaks with certain providers while using the Proxy param. >> >> - The Request URI in the REG/INV will reflect the Proxy param. >> - The Proxy-Auth headers has a URI parameter that will now have the >> Request URI ( Proxy param). >> >> Some SIP providers have the Realm/Domain different from the Proxy. So >> they end up rejecting the request since the domain in the proxy is not in >> their white list of authenticated hosts. >> >> Eg. Proxy - sip10.voip.xyz.com >> Realm - xyz.com >> >> The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy >> Auth header will also have the same URI. The Providers I know are unwilling >> to add the Proxy domain sip10.voip.xyz.com into their white-list of >> authenticated hosts. They require the Proxy-Auth to have xyz.com. >> >> I am not sure if I made myself clear ? >> >> Anyway, I managed to set the sip_invite_route_uri as the Proxy. That >> seemed to work even for gateways. >> >> Thank you, >> Sharath >> >> >> On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: >> >>> Why not use proxy param in gateway? >>> >>> On 19 Oct 2017 3:11 am, "Sharath Kumar" >>> wrote: >>> >>>> Can we add an fs_path when dialing out via a gateway ? I know this has >>>> been asked before and in the archives it has been said to support and also >>>> not support fs_path in a gateway. Just wanted to confirm. >>>> >>>> thank you, >>>> Shaks. >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 shakumarsoftware at gmail.com Wed Oct 25 17:02:54 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Wed, 25 Oct 2017 11:02:54 -0600 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: Srigo, You are right. The oubound-proxy works fine too. The only problem I have is it is global per profile. My gateways are defined in my SIP Profiles. So what if I have 2 gateways each having different outbound-proxies :) ? Thank you, Shaks On Wed, Oct 25, 2017 at 1:20 AM, Srigo Kanapathipillai wrote: > Hi Sharath Kumar, > > Use the params: outbound-proxy and proxy. > > Eg: > > Outbound-proxy: sip10.voip.xyz.com > Proxy: xyz.com > > Srigo > > Sent from my iPhone > > On 24 Oct 2017, at 19:04, Sharath Kumar > wrote: > > Well, that is an option but I found out that the SIP Authentication breaks > with certain providers while using the Proxy param. > > - The Request URI in the REG/INV will reflect the Proxy param. > - The Proxy-Auth headers has a URI parameter that will now have the > Request URI ( Proxy param). > > Some SIP providers have the Realm/Domain different from the Proxy. So they > end up rejecting the request since the domain in the proxy is not in their > white list of authenticated hosts. > > Eg. Proxy - sip10.voip.xyz.com > Realm - xyz.com > > The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy > Auth header will also have the same URI. The Providers I know are unwilling > to add the Proxy domain sip10.voip.xyz.com into their white-list of > authenticated hosts. They require the Proxy-Auth to have xyz.com. > > I am not sure if I made myself clear ? > > Anyway, I managed to set the sip_invite_route_uri as the Proxy. That > seemed to work even for gateways. > > Thank you, > Sharath > > > On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: > >> Why not use proxy param in gateway? >> >> On 19 Oct 2017 3:11 am, "Sharath Kumar" >> wrote: >> >>> Can we add an fs_path when dialing out via a gateway ? I know this has >>> been asked before and in the archives it has been said to support and also >>> not support fs_path in a gateway. Just wanted to confirm. >>> >>> thank you, >>> Shaks. >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 aphoticwalker at gmail.com Thu Oct 26 02:22:44 2017 From: aphoticwalker at gmail.com (Charles Yu) Date: Thu, 26 Oct 2017 10:22:44 +0800 Subject: [Freeswitch-users] Control register behavior via a lua script when REGISTER? In-Reply-To: References: Message-ID: Hi Michael, Thank you for your reply. I will try that. 2017-10-12 21:48 GMT+08:00 Michael Jerris : > yes, lua directory hooks. > > On Thu, Oct 12, 2017 at 4:23 AM Charles Yu > wrote: > >> Hi, >> >> Is there a way to control register behavior via a lua script when >> receiving sip REGISTER ? Such like: I can reject this REGISTER if it does >> not match some condition. Trigger a sofia::register_failure event? Any >> ideas? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 servtelar at gmail.com Thu Oct 26 02:32:26 2017 From: servtelar at gmail.com (GM phy) Date: Wed, 25 Oct 2017 23:32:26 -0300 Subject: [Freeswitch-users] ciphers not available for verto wss (neither wss on sip profile) Message-ID: Hello, im having this issue and by now i found no way to figure it out. enabling tls on internal, if a exec openssl s_client -connect 10.10.10.9:5061 -tls1_2 Connection is established and server returns my certificate and this information -- New, TLSv1/SSLv3, Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-ECDSA-AES256-GCM-SHA384 ... -- When i run the same comand for port 7443 (the wss binding on internal profile) or port 8082 (verto wss) i get (same on both cases) -- New, TLSv1/SSLv3, Cipher is ECDH-RSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDH-RSA-AES256-GCM-SHA384 ... -- This causes that all of the webrtc phones we're using (some with sipjs, some with verto) are unable to establish the wss socket in order to register and make calls. On the browser console i get Protocol mismatch, and on a pcap, server rejects the ssl handshake with code 40. I've been working with the code on mod_verto.c and ws.c with no luck. basically what im looking for is for verto and/or wss on internal profile, to support Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 (or ECDHE-ECDSA) in order to webrtc can connect to fs. any help will be really appreciated. thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From ksrigo at gmail.com Thu Oct 26 07:53:51 2017 From: ksrigo at gmail.com (KSrigo) Date: Thu, 26 Oct 2017 09:53:51 +0200 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: References: Message-ID: <5479127B-E572-409A-8015-C6F663CA12BB@gmail.com> Hi Sharath, You have to do it the following way: 1. Create the SIP profile (it will have 2 or more gateways). DO not add gateways conf inside the SIP Profiles. - On the top of the SIP profile add this: 2. Inside the folder sip_profiles create the folder gateways and add your 2 gateways config (1 file per provider) —> Here you can define different outboud- proxy per gateway 3. Restart/Reload your sip_profiles 4. From fs_cli, if you do 'sofia status gateway’, you should be able to see your both gateways and status. Good luck Srigo > On Oct 25, 2017, at 7:02 PM, Sharath Kumar wrote: > > Srigo, > > You are right. The oubound-proxy works fine too. The only problem I have is it is global per profile. My gateways are defined in my SIP Profiles. So what if I have 2 gateways each having different outbound-proxies :) ? > > Thank you, > Shaks > > On Wed, Oct 25, 2017 at 1:20 AM, Srigo Kanapathipillai > wrote: > Hi Sharath Kumar, > > Use the params: outbound-proxy and proxy. > > Eg: > > Outbound-proxy: sip10.voip.xyz.com > Proxy: xyz.com > > Srigo > > Sent from my iPhone > > On 24 Oct 2017, at 19:04, Sharath Kumar > wrote: > >> Well, that is an option but I found out that the SIP Authentication breaks with certain providers while using the Proxy param. >> >> - The Request URI in the REG/INV will reflect the Proxy param. >> - The Proxy-Auth headers has a URI parameter that will now have the Request URI ( Proxy param). >> >> Some SIP providers have the Realm/Domain different from the Proxy. So they end up rejecting the request since the domain in the proxy is not in their white list of authenticated hosts. >> >> Eg. Proxy - sip10.voip.xyz.com >> Realm - xyz.com >> >> The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy Auth header will also have the same URI. The Providers I know are unwilling to add the Proxy domain sip10.voip.xyz.com into their white-list of authenticated hosts. They require the Proxy-Auth to have xyz.com . >> >> I am not sure if I made myself clear ? >> >> Anyway, I managed to set the sip_invite_route_uri as the Proxy. That seemed to work even for gateways. >> >> Thank you, >> Sharath >> >> >> On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas > wrote: >> Why not use proxy param in gateway? >> >> On 19 Oct 2017 3:11 am, "Sharath Kumar" > wrote: >> Can we add an fs_path when dialing out via a gateway ? I know this has been asked before and in the archives it has been said to support and also not support fs_path in a gateway. Just wanted to confirm. >> >> thank you, >> Shaks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 ksrigo at gmail.com Thu Oct 26 08:01:33 2017 From: ksrigo at gmail.com (KSrigo) Date: Thu, 26 Oct 2017 10:01:33 +0200 Subject: [Freeswitch-users] outbound route question In-Reply-To: <3e21ec25.5359.15f2d95a327.Coremail.hiastar_alex@163.com> References: <3e21ec25.5359.15f2d95a327.Coremail.hiastar_alex@163.com> Message-ID: <87B203B4-0BA2-498B-8CAA-34DB156A0079@gmail.com> Hi, 1. It seems your call is catched by another dialplan extension. Make sure once you transfer the call to the default context, your new extension is on the top of everything. 2. > > > > > > > > But it didn't take the route. > Why is that? what should i do? > This are CLI log: > 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000 at 192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public > 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000 at 192.168.1.70 to XML[1001 at default] > 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. > 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features > 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features > 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001 at 192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] > 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001 at 192.168.1.69:5060! > 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media > 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000 at 192.168.1.70! > 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000 at 192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001 at 192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] > 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001 at 192.168.1.69:5060) Ended > 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001 at 192.168.1.69:5060 [CS_DESTROY] > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000 at 192.168.1.70) Ended > 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000 at 192.168.1.70 [CS_DESTROY] > > Thanks for answer. > > > 【网易自营|30天无忧退货】仅售同款价1/4！MUJI制造商“2017秋冬舒适家居拖鞋系列”限时仅34.9元>>   > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 khamlichi.khalil at gmail.com Thu Oct 26 08:05:23 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Thu, 26 Oct 2017 09:05:23 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: The customer is billed by a third party (by his own telephone operator on his phone bill) as soon as the call is established, so at the call center side the need is to be able to not answer the phone until an agent is available, if it times-out, its ok, maybe the call center should provide more agents or something but it still is an issue that is not to manage by the software, what I noticed though is that their system is already configured for fail-over to some other location after 8 seconds of ringing, so the requirement to not answer is already taken care of upstream . let me ask a new question instead : how can I put a call on hold without answering ? If I can do this I will write some code that will check availability of agents and only then put call in queue. Thanks a lot for your help On Wed, Oct 25, 2017 at 2:40 PM, GM phy wrote: > You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. > > >> On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: >> >> In our case it is really mandatory, if customer does not get to talk >> to an agent the company has no legal right to bill him anything. for >> this we really must implement waiting without answering the call, >> >> On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >>> Don't do that. :) >>> >>> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>>> >>>> Hi >>>> >>>> That’s not a mandatory requirement at all, waiting time counts as part of >>>> the call. >>>> Doing something like that you’ll have a network disconnection due lack of >>>> answer, no matter if you send 180 or 183. The timeout will depend on the >>>> scenario on each country, but in general terms it will be between one or two >>>> minutes. >>>> >>>> GM >>>> >>>> >>>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>>> wrote: >>>>> >>>>> Hi, >>>>> >>>>> is it possible to use mod_callcenter queues without answering the >>>>> line, that is the customer is held in the queue and hearing ringing >>>>> (without the call being answered). >>>>> >>>>> In fact this is a mandatory requirement when calls have a cost and we >>>>> don't want to charge customer for waiting time but only for talk time. >>>>> >>>>> Thanks in advance. >>>>> >>>>> Khalil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From khamlichi.khalil at gmail.com Thu Oct 26 08:25:59 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Thu, 26 Oct 2017 09:25:59 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> Message-ID: ok I found it, I used uuid_ring_ready , it is providing ringing back to customer with no answering so no billing on operator side, I can take any time I want to check availability of agent. Thanks again to all of you. On Thu, Oct 26, 2017 at 9:05 AM, Khalil Khamlichi wrote: > The customer is billed by a third party (by his own telephone operator > on his phone bill) as soon as the call is established, so at the call > center side the need is to be able to not answer the phone until an > agent is available, if it times-out, its ok, maybe the call center > should provide more agents or something but it still is an issue that > is not to manage by the software, what I noticed though is that their > system is already configured for fail-over to some other location > after 8 seconds of ringing, so the requirement to not answer is > already taken care of upstream > . > let me ask a new question instead : > how can I put a call on hold without answering ? If I can do this I > will write some code that will check availability of agents and only > then put call in queue. > > Thanks a lot for your help > > On Wed, Oct 25, 2017 at 2:40 PM, GM phy wrote: >> You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. >> >> >>> On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: >>> >>> In our case it is really mandatory, if customer does not get to talk >>> to an agent the company has no legal right to bill him anything. for >>> this we really must implement waiting without answering the call, >>> >>> On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >>>> Don't do that. :) >>>> >>>> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>>>> >>>>> Hi >>>>> >>>>> That’s not a mandatory requirement at all, waiting time counts as part of >>>>> the call. >>>>> Doing something like that you’ll have a network disconnection due lack of >>>>> answer, no matter if you send 180 or 183. The timeout will depend on the >>>>> scenario on each country, but in general terms it will be between one or two >>>>> minutes. >>>>> >>>>> GM >>>>> >>>>> >>>>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>>>> wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> is it possible to use mod_callcenter queues without answering the >>>>>> line, that is the customer is held in the queue and hearing ringing >>>>>> (without the call being answered). >>>>>> >>>>>> In fact this is a mandatory requirement when calls have a cost and we >>>>>> don't want to charge customer for waiting time but only for talk time. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> Khalil >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From shaun.stokes at itec-support.co.uk Thu Oct 26 08:42:56 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 26 Oct 2017 08:42:56 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> , Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F6EECD85@mbx-01.sysconfig.co.uk> To put the call on hold I believe it would need to be answered as I'm fairly sure you can't have the hold state while the call is ringing. The ideal solution would be for you to send them the call logs which show the amount of time the agent has spent on the phone with the caller rather than using the telephone operators bill. If this isn't possible then another option is to develop call centre module to have the option to keep the call in the ringing state until the agent has answered, I'm not familiar enough with the source code to point you in the right direction and this would also create hoops to jump through when you want to update FreeSWITCH (unless of course this feature gets added to the project). ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Khalil Khamlichi [khamlichi.khalil at gmail.com] Sent: 26 October 2017 09:05 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line The customer is billed by a third party (by his own telephone operator on his phone bill) as soon as the call is established, so at the call center side the need is to be able to not answer the phone until an agent is available, if it times-out, its ok, maybe the call center should provide more agents or something but it still is an issue that is not to manage by the software, what I noticed though is that their system is already configured for fail-over to some other location after 8 seconds of ringing, so the requirement to not answer is already taken care of upstream . let me ask a new question instead : how can I put a call on hold without answering ? If I can do this I will write some code that will check availability of agents and only then put call in queue. Thanks a lot for your help On Wed, Oct 25, 2017 at 2:40 PM, GM phy wrote: > You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. > > >> On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: >> >> In our case it is really mandatory, if customer does not get to talk >> to an agent the company has no legal right to bill him anything. for >> this we really must implement waiting without answering the call, >> >> On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >>> Don't do that. :) >>> >>> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>>> >>>> Hi >>>> >>>> That’s not a mandatory requirement at all, waiting time counts as part of >>>> the call. >>>> Doing something like that you’ll have a network disconnection due lack of >>>> answer, no matter if you send 180 or 183. The timeout will depend on the >>>> scenario on each country, but in general terms it will be between one or two >>>> minutes. >>>> >>>> GM >>>> >>>> >>>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>>> wrote: >>>>> >>>>> Hi, >>>>> >>>>> is it possible to use mod_callcenter queues without answering the >>>>> line, that is the customer is held in the queue and hearing ringing >>>>> (without the call being answered). >>>>> >>>>> In fact this is a mandatory requirement when calls have a cost and we >>>>> don't want to charge customer for waiting time but only for talk time. >>>>> >>>>> Thanks in advance. >>>>> >>>>> Khalil >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From shaun.stokes at itec-support.co.uk Thu Oct 26 08:46:16 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 26 Oct 2017 08:46:16 +0000 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> , Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A902F6EECDA3@mbx-01.sysconfig.co.uk> Glad you found a solution, I'm not familiar with uuid_ring_ready, this would be worth adding this to the FreeSWITCH documentation. ________________________________________ From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Khalil Khamlichi [khamlichi.khalil at gmail.com] Sent: 26 October 2017 09:25 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line ok I found it, I used uuid_ring_ready , it is providing ringing back to customer with no answering so no billing on operator side, I can take any time I want to check availability of agent. Thanks again to all of you. On Thu, Oct 26, 2017 at 9:05 AM, Khalil Khamlichi wrote: > The customer is billed by a third party (by his own telephone operator > on his phone bill) as soon as the call is established, so at the call > center side the need is to be able to not answer the phone until an > agent is available, if it times-out, its ok, maybe the call center > should provide more agents or something but it still is an issue that > is not to manage by the software, what I noticed though is that their > system is already configured for fail-over to some other location > after 8 seconds of ringing, so the requirement to not answer is > already taken care of upstream > . > let me ask a new question instead : > how can I put a call on hold without answering ? If I can do this I > will write some code that will check availability of agents and only > then put call in queue. > > Thanks a lot for your help > > On Wed, Oct 25, 2017 at 2:40 PM, GM phy wrote: >> You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. >> >> >>> On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: >>> >>> In our case it is really mandatory, if customer does not get to talk >>> to an agent the company has no legal right to bill him anything. for >>> this we really must implement waiting without answering the call, >>> >>> On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >>>> Don't do that. :) >>>> >>>> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>>>> >>>>> Hi >>>>> >>>>> That’s not a mandatory requirement at all, waiting time counts as part of >>>>> the call. >>>>> Doing something like that you’ll have a network disconnection due lack of >>>>> answer, no matter if you send 180 or 183. The timeout will depend on the >>>>> scenario on each country, but in general terms it will be between one or two >>>>> minutes. >>>>> >>>>> GM >>>>> >>>>> >>>>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>>>> wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> is it possible to use mod_callcenter queues without answering the >>>>>> line, that is the customer is held in the queue and hearing ringing >>>>>> (without the call being answered). >>>>>> >>>>>> In fact this is a mandatory requirement when calls have a cost and we >>>>>> don't want to charge customer for waiting time but only for talk time. >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> Khalil >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From akshaya18j at gmail.com Thu Oct 26 11:41:07 2017 From: akshaya18j at gmail.com (Akshaya j) Date: Thu, 26 Oct 2017 17:11:07 +0530 Subject: [Freeswitch-users] [ERR] mod_lua.cpp:103 attempt to call a nil value in recordFile function Message-ID: Hi team, Spec: OS - Debian FS server Version -1.6.15 I use Zoiper clients (in Ubuntu machine) directly connected to FS server. I'm supposed to write voicemail scripts where i get the call and bridge to Avaya gateway if there is no response i load a lua script. The lua script contains........... if session:ready() then session:speak("Record your message now, to end recording press pound"); session:setInputCallback('onInput',''); test = session:recordFile(recording_filename, 60, 30, 5) session:consoleLog("info", "session:recordFile() = " .. test ) And my Input CallBack function has got..... function onInput(session, input_type, data) session:consoleLog("info", "inside" ) if input_type == "dtmf" then return "break" end end Now my problem is I dont see any log which is inside the callback function.I can record the voice and getting saved but there is an error.... [DEBUG] switch_rtp.c:7255 RTP RECV DTMF #:1280 [INFO] switch_channel.c:515 RECV DTMF #:1280 [ERR] mod_lua.cpp:103 attempt to call a nil value stack traceback: [C]: in function 'recordFile' /usr/local/freeswitch/scripts/test.lua:19: in main chunk I can witness that the session is receiving the DTMF signal but then the function is not getting executed. I tried commenting the func call of callback and I can witness the DTMF is not sensed and hence the record termination happens only after silence_threshold. I badly need to process the keypress DTMF for more options without console err. NOTE: instead of recordFile() I tried speak() even by then i get the same error "attempt to call a nil value" in speak function. Please save me FS.! -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Thu Oct 26 11:44:53 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Thu, 26 Oct 2017 12:44:53 +0100 Subject: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A902F6EECDA3@mbx-01.sysconfig.co.uk> References: <8B33B1A4-AB40-463A-B355-0904ECFB66BD@gmail.com> <6FD2F8B5BB72834E9939AEDF9FB802A902F6EECDA3@mbx-01.sysconfig.co.uk> Message-ID: Honestly since we switched to Freeswitch, average time for resolution of problems does not go beyond 2 days, sometimes couple minutes on the documentation, with other solutions, we were always stuck for months and months. On Thu, Oct 26, 2017 at 9:46 AM, Shaun Stokes wrote: > Glad you found a solution, I'm not familiar with uuid_ring_ready, this would be worth adding this to the FreeSWITCH documentation. > ________________________________________ > From: FreeSWITCH-users [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Khalil Khamlichi [khamlichi.khalil at gmail.com] > Sent: 26 October 2017 09:25 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] is it possible to use mod_callcenter queues without answering the line > > ok I found it, > > I used uuid_ring_ready , it is providing ringing back to customer > with no answering so no billing on operator side, I can take any time > I want to check availability of agent. > > Thanks again to all of you. > > On Thu, Oct 26, 2017 at 9:05 AM, Khalil Khamlichi > wrote: >> The customer is billed by a third party (by his own telephone operator >> on his phone bill) as soon as the call is established, so at the call >> center side the need is to be able to not answer the phone until an >> agent is available, if it times-out, its ok, maybe the call center >> should provide more agents or something but it still is an issue that >> is not to manage by the software, what I noticed though is that their >> system is already configured for fail-over to some other location >> after 8 seconds of ringing, so the requirement to not answer is >> already taken care of upstream >> . >> let me ask a new question instead : >> how can I put a call on hold without answering ? If I can do this I >> will write some code that will check availability of agents and only >> then put call in queue. >> >> Thanks a lot for your help >> >> On Wed, Oct 25, 2017 at 2:40 PM, GM phy wrote: >>> You said it all, “in our case”, but the network works in a different way. So as Shaun pointed out, you can use the agent leg to bill your customer. >>> >>> >>>> On Oct 25, 2017, at 3:32 AM, Khalil Khamlichi wrote: >>>> >>>> In our case it is really mandatory, if customer does not get to talk >>>> to an agent the company has no legal right to bill him anything. for >>>> this we really must implement waiting without answering the call, >>>> >>>> On Tue, Oct 24, 2017 at 11:36 PM, Ítalo Rossi wrote: >>>>> Don't do that. :) >>>>> >>>>> Em ter, 24 de out de 2017 às 18:16, GM phy escreveu: >>>>>> >>>>>> Hi >>>>>> >>>>>> That’s not a mandatory requirement at all, waiting time counts as part of >>>>>> the call. >>>>>> Doing something like that you’ll have a network disconnection due lack of >>>>>> answer, no matter if you send 180 or 183. The timeout will depend on the >>>>>> scenario on each country, but in general terms it will be between one or two >>>>>> minutes. >>>>>> >>>>>> GM >>>>>> >>>>>> >>>>>>> On Oct 24, 2017, at 4:00 PM, Khalil Khamlichi >>>>>>> wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> is it possible to use mod_callcenter queues without answering the >>>>>>> line, that is the customer is held in the queue and hearing ringing >>>>>>> (without the call being answered). >>>>>>> >>>>>>> In fact this is a mandatory requirement when calls have a cost and we >>>>>>> don't want to charge customer for waiting time but only for talk time. >>>>>>> >>>>>>> Thanks in advance. >>>>>>> >>>>>>> Khalil >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus Scanning Service. > ______________________________________________________________________ > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jose.lopes at itcenter.com.pt Thu Oct 26 13:16:42 2017 From: jose.lopes at itcenter.com.pt (=?UTF-8?Q?Jos=C3=A9_Lopes?=) Date: Thu, 26 Oct 2017 14:16:42 +0100 Subject: [Freeswitch-users] Share Screen stop work after ~20 seconds (FS 1.6.18/1.6.19/master) In-Reply-To: References: Message-ID: Hello, Thanks for your reply. I created an issue at https://freeswitch.org/jira/browse/FS-10759 . Thanks and Best Regards, Jose Lopes On Thu, Oct 19, 2017 at 10:52 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Please report issues to jira not the mailing list. > > On Thu, Oct 19, 2017 at 10:11 AM, José Lopes > wrote: > >> Hello, >> >> I have a Freeswitch WebRTC SIP connected with a Freeswitch MCU. >> >> When i use Video Conference on WebRTC, when I share the Screen, the >> screen is shared, but after 20 seconds the screen freeze and it stops to be >> presented. >> The video call of Screen Share is still active. >> >> I have this situation after upgrade to FreeSwitch 1.6.18. >> On Freeswitch 1.6.17 this situation does not happen. >> I tried also on Freeswitch 1.6.19 and master and this situation still >> happens. >> >> There is any configuration that I need to use on the new versions related >> to this scenario, so this will not happen? >> >> >> Additional Notes: >> * On Freeswitch MCU, I made a change that when it is video from Screen >> Share, the conference member is with flags : mute and deaf. After 20 >> seconds the video freeze , but after some seconds the screen share start >> working again. >> * I tried a Freeswitch WebRTC with MCU, and this situation doesn't happen. >> >> >> Thanks in advance >> >> >> >> Best regards, >> >> *José Lopes* >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> > > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From marko.mrvelj at amphinicy.com Wed Oct 25 11:22:54 2017 From: marko.mrvelj at amphinicy.com (Marko Mrvelj) Date: Wed, 25 Oct 2017 13:22:54 +0200 (CEST) Subject: [Freeswitch-users] Adding authorization to SIP messaging In-Reply-To: <112898af-eb13-7aec-af47-82e911399bb8@wirelessmundi.com> References: <020f01d34d75$581b1a80$08514f80$@amphinicy.com> <112898af-eb13-7aec-af47-82e911399bb8@wirelessmundi.com> Message-ID: <024c01d34d83$cf564b70$6e02e250$@amphinicy.com> I can confirm it works as expected, thanks! -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of António Silva Sent: Wednesday, October 25, 2017 12:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Adding authorization to SIP messaging Hi, yes, you can set in the sofia profile the parameter: On 10/25/2017 11:39 AM, Marko Mrvelj wrote: > Hi all, > > We are experimenting with chatplan, in order to get basic chat > functionality. > > Messages do work as expected, but there is a problem that message can > be submitted to server without any authorization. Could somebody tell > me if it is possible to restrict messages to only authorized users? > How can this be done? > > Thanks in advance, > Marko > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Saludos / Regards / Cumprimentos António Silva _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From richard.screene at thisisdrum.com Thu Oct 26 15:19:17 2017 From: richard.screene at thisisdrum.com (Richard Screene) Date: Thu, 26 Oct 2017 16:19:17 +0100 Subject: [Freeswitch-users] mod_conference video_no_video_avatar_png flag with video-mode passthrough Message-ID: <1ECCEE05-8527-40DC-B219-705C5720AE82@thisisdrum.com> Hello, In mod_conference, is it possible to use the video_no_video_avatar_png flag with video-mode passthrough? I can’t get it working but that could indicate that either (a) its not possible or (b) I’ve done something wrong. What I’m trying to achieve is: in a conference with both audio/video and audio only members the video stream follows the active speaker. However, if one of the audio only participants is speaking then a placeholder image is transmitted to the audio/video members instead. Obviously, I’d prefer to reduce the amount of transcoding/canvas manipulation on the server hence the use of the passthrough video-mode. If there any better ways of achieving this then please let me know! Many thanks for reading this, Richard From emdevane at gmail.com Thu Oct 26 21:03:59 2017 From: emdevane at gmail.com (Erik M. Devane - Comms Guy) Date: Thu, 26 Oct 2017 16:03:59 -0500 Subject: [Freeswitch-users] Multiple "Changing expire time to 60 by request of proxy" requests result in delayed audio Message-ID: Hello, I have a fairly generic upstream VoIP provider. Every day or so of calls, I receive: 2017-10-26 16:54:06.100048 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:54:31.120584 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:09.139687 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:48.160453 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:04.180054 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:26.180559 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:43.199818 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:57:24.219901 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:08.260016 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:29.259860 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:59:00.281354 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 If there are just one or two, nothing seems to happen. When I see ten or more, there's an audio delay of five seconds or so. Can anyone help point me in the right direction? Thank you, Erik -------------- next part -------------- An HTML attachment was scrubbed... URL: From gb at cm.nl Fri Oct 27 07:45:40 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 27 Oct 2017 07:45:40 +0000 Subject: [Freeswitch-users] 180/183 not passed through sip profiles Message-ID: Hi, We're using a freeswitch which bridges calls between two networks: private and public. The server has two interfaces. The calls originate from the private network and are bridged using the public sip profile. For some reason I cannot get the 180 and 183 messages passed back from the outbound leg to the inbound leg when sending calls over the public sip profile. Both sip profiles have been set with: The below dialplan is hit when a call arrives on the LAN(private) interface of the freeswitch. Any ideas why the 180/183 are not relayed from outbound leg back to inbound leg? A trace shows these messages do arrive on the outbound leg. Grant Bagdasarian Developer +31765727054 cm.com [cid:image001.png at 01D34F08.7CB33B90] -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: image001.png URL: From alexandr.popov at iqoption.com Fri Oct 27 09:14:53 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Fri, 27 Oct 2017 12:14:53 +0300 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: because freeswitch is B2BUA i guess) 2017-10-27 10:45 GMT+03:00 Grant Bagdasarian : > Hi, > > > > We’re using a freeswitch which bridges calls between two networks: private > and public. The server has two interfaces. The calls originate from the > private network and are bridged using the public sip profile. > > For some reason I cannot get the 180 and 183 messages passed back from the > outbound leg to the inbound leg when sending calls over the public sip > profile. > > > > Both sip profiles have been set with: > > > > > > > > The below dialplan is hit when a call arrives on the LAN(private) > interface of the freeswitch. > > > > > > > > > > > > require-nested="false"> > > > > > > > > > > > > > > > > > > > > > > /> > > > > > > > > > > > > Any ideas why the 180/183 are not relayed from outbound leg back to > inbound leg? A trace shows these messages do arrive on the outbound leg. > > > > Grant Bagdasarian > > Developer > > +31765727054 > > cm.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: not available URL: From mirkobrankovic at gmail.com Fri Oct 27 09:29:59 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Fri, 27 Oct 2017 11:29:59 +0200 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: Hi, Have you tried with bridge_early_media=false, since in pproxy_mode everything that has to do with live media will fail? --mirko On Fri, Oct 27, 2017 at 9:45 AM, Grant Bagdasarian wrote: > Hi, > > > > We’re using a freeswitch which bridges calls between two networks: private > and public. The server has two interfaces. The calls originate from the > private network and are bridged using the public sip profile. > > For some reason I cannot get the 180 and 183 messages passed back from the > outbound leg to the inbound leg when sending calls over the public sip > profile. > > > > Both sip profiles have been set with: > > > > > > > > The below dialplan is hit when a call arrives on the LAN(private) > interface of the freeswitch. > > > > > > > > > > > > require-nested="false"> > > > > > > > > > > > > > > > > > > > > > > /> > > > > > > > > > > > > Any ideas why the 180/183 are not relayed from outbound leg back to > inbound leg? A trace shows these messages do arrive on the outbound leg. > > > > Grant Bagdasarian > > Developer > > +31765727054 > > cm.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: not available URL: From gb at cm.nl Fri Oct 27 09:36:01 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 27 Oct 2017 09:36:01 +0000 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: Hi Mirko, I’ll try that. To answer Alexandr Popov’s comment, we also have a FS instance running with just a single sip profile that also does a bridge to an endpoint on the internet, but uses SIP-ALG provided by our firewall, which works just fine. Not sure if this is related to multiple sip profiles. That’s the only difference between the two. Regards, Grant From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mirko Brankovic Sent: vrijdag 27 oktober 2017 11:30 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 180/183 not passed through sip profiles Hi, Have you tried with bridge_early_media=false, since in pproxy_mode everything that has to do with live media will fail? --mirko On Fri, Oct 27, 2017 at 9:45 AM, Grant Bagdasarian > wrote: Hi, We’re using a freeswitch which bridges calls between two networks: private and public. The server has two interfaces. The calls originate from the private network and are bridged using the public sip profile. For some reason I cannot get the 180 and 183 messages passed back from the outbound leg to the inbound leg when sending calls over the public sip profile. Both sip profiles have been set with: The below dialplan is hit when a call arrives on the LAN(private) interface of the freeswitch. Any ideas why the 180/183 are not relayed from outbound leg back to inbound leg? A trace shows these messages do arrive on the outbound leg. Grant Bagdasarian Developer +31765727054 cm.com [cid:image001.png at 01D34F17.E72D3CA0] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: image001.png URL: From gb at cm.nl Fri Oct 27 09:39:41 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 27 Oct 2017 09:39:41 +0000 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: That’s actually not true, the other FS instance uses bypass-media true. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian Sent: vrijdag 27 oktober 2017 11:36 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 180/183 not passed through sip profiles Hi Mirko, I’ll try that. To answer Alexandr Popov’s comment, we also have a FS instance running with just a single sip profile that also does a bridge to an endpoint on the internet, but uses SIP-ALG provided by our firewall, which works just fine. Not sure if this is related to multiple sip profiles. That’s the only difference between the two. Regards, Grant From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mirko Brankovic Sent: vrijdag 27 oktober 2017 11:30 To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] 180/183 not passed through sip profiles Hi, Have you tried with bridge_early_media=false, since in pproxy_mode everything that has to do with live media will fail? --mirko On Fri, Oct 27, 2017 at 9:45 AM, Grant Bagdasarian > wrote: Hi, We’re using a freeswitch which bridges calls between two networks: private and public. The server has two interfaces. The calls originate from the private network and are bridged using the public sip profile. For some reason I cannot get the 180 and 183 messages passed back from the outbound leg to the inbound leg when sending calls over the public sip profile. Both sip profiles have been set with: The below dialplan is hit when a call arrives on the LAN(private) interface of the freeswitch. Any ideas why the 180/183 are not relayed from outbound leg back to inbound leg? A trace shows these messages do arrive on the outbound leg. Grant Bagdasarian Developer +31765727054 cm.com [cid:image001.png at 01D34F18.69EEBF60] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: image001.png URL: From achinthau at gmail.com Fri Oct 27 11:36:18 2017 From: achinthau at gmail.com (Achintha) Date: Fri, 27 Oct 2017 17:06:18 +0530 Subject: [Freeswitch-users] Freeswitch crashed daily Message-ID: hi all Our freeswitch (Version 1.6.18-35-6e79667~64bit) handled around 7000 calls per day. Its coredb configured with postgresql (bdr) database (freeswitch DB) and following modules. mod_xml_curl : Dynamic Dialplans (from rest Service) mod_json_cdr : for CDR (from rest Service) and one module developed by our self (mod_ards) freeswitch crashed daily. here i have upload a core dump trace on pastbin https://pastebin.freeswitch.org/view/bf2b479d -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: From jurijs.ivolga at gmail.com Fri Oct 27 11:53:05 2017 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Fri, 27 Oct 2017 14:53:05 +0300 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: Hi, Not sure that it will help, but I think something similar was discussed here: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-May/083985.html I would recommend you to set ignore_early_media inside bridge something like this: Additionally you can try to send 180 or 183 for all calls or execute it based on event(please check freeswitch events), never tried though, but it may work. Off cause you can try to move to one profile only... With kind regards, Jurijs On Fri, Oct 27, 2017 at 12:39 PM, Grant Bagdasarian wrote: > That’s actually not true, the other FS instance uses bypass-media true. > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Grant Bagdasarian > *Sent:* vrijdag 27 oktober 2017 11:36 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] 180/183 not passed through sip profiles > > > > Hi Mirko, > > > > I’ll try that. > > > > To answer Alexandr Popov’s comment, we also have a FS instance running > with just a single sip profile that also does a bridge to an endpoint on > the internet, but uses SIP-ALG provided by our firewall, which works just > fine. Not sure if this is related to multiple sip profiles. That’s the only > difference between the two. > > > > Regards, > > > > Grant > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Mirko > Brankovic > *Sent:* vrijdag 27 oktober 2017 11:30 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] 180/183 not passed through sip profiles > > > > Hi, > > Have you tried with bridge_early_media=false, since in pproxy_mode > everything that has to do with live media will fail? > > > > --mirko > > > > On Fri, Oct 27, 2017 at 9:45 AM, Grant Bagdasarian wrote: > > Hi, > > > > We’re using a freeswitch which bridges calls between two networks: private > and public. The server has two interfaces. The calls originate from the > private network and are bridged using the public sip profile. > > For some reason I cannot get the 180 and 183 messages passed back from the > outbound leg to the inbound leg when sending calls over the public sip > profile. > > > > Both sip profiles have been set with: > > > > > > > > The below dialplan is hit when a call arrives on the LAN(private) > interface of the freeswitch. > > > > > > > > > > > > require-nested="false"> > > > > > > > > > > > > > > > > > > > > > > /> > > > > > > > > > > > > Any ideas why the 180/183 are not relayed from outbound leg back to > inbound leg? A trace shows these messages do arrive on the outbound leg. > > > > Grant Bagdasarian > > Developer > > +31765727054 > > cm.com > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > Regards, > > Mirko > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: not available URL: From nico at vthadden.de Fri Oct 27 17:19:23 2017 From: nico at vthadden.de (Nicola von Thadden) Date: Fri, 27 Oct 2017 19:19:23 +0200 Subject: [Freeswitch-users] Gigaset auto answer In-Reply-To: <749EFE28-B29C-4545-8211-5F86F205EC54@vallimamod.org> References: <1F7FD689-BB26-4105-8532-159131C5E4D6@vallimamod.org> <749EFE28-B29C-4545-8211-5F86F205EC54@vallimamod.org> Message-ID: HI, yes, the to domain is the right one. Defining the variables in the dialplan works as expected, FS authenticates to the phone when requested and intercom is working. Is there anything I can do to help solve the other implementation with the reverse lookup? Thanks for your help Nico On 10/24/2017 03:16 PM, Vallimamod Abdullah wrote: > Hi, > > It looks like the reverse lookup didn't work as expected. Is the To domain from the 401 reply the same as your $${domain}? > You can also define sip_auth_username and sip_auth_password channel variables in the dialplan, they will be used for the reverse lookup. > > About the double quotation mark, the source code shows that the realm value is not unquoted when retrieved from the Authentication header. It looks like the sofia lib requires it so to compute the hash. > > Best Regards From vma at vallimamod.org Fri Oct 27 21:24:14 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 27 Oct 2017 23:24:14 +0200 Subject: [Freeswitch-users] Multiple "Changing expire time to 60 by request of proxy" requests result in delayed audio In-Reply-To: References: Message-ID: <199D32EE-760A-4E8F-B8A3-63EFEAFD7A35@vallimamod.org> Hi, I don't think it's related. The log lines are about the registration expiry: your provider is imposing 60s expiry time, which is different from what is defined in your config, so freeswitch adjusts the value. You can mute it by setting the expire-seconds param to 60 in your gateway config. Audio delays are generally related to system load or network issues resulting in jitter or packet losses. You can make a mtr and a tcpdump at different points and analyse the result. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 26 Oct 2017, at 23:03, Erik M. Devane - Comms Guy wrote: > > Hello, > > I have a fairly generic upstream VoIP provider. Every day or so of calls, I receive: > > 2017-10-26 16:54:06.100048 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:54:31.120584 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:55:09.139687 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:55:48.160453 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:56:04.180054 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:56:26.180559 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:56:43.199818 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:57:24.219901 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:58:08.260016 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:58:29.259860 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > 2017-10-26 16:59:00.281354 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 > > If there are just one or two, nothing seems to happen. > > When I see ten or more, there's an audio delay of five seconds or so. > > Can anyone help point me in the right direction? > > Thank you, > > Erik > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shakumarsoftware at gmail.com Fri Oct 27 22:32:16 2017 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Fri, 27 Oct 2017 16:32:16 -0600 Subject: [Freeswitch-users] FS_PATH for Gateways In-Reply-To: <5479127B-E572-409A-8015-C6F663CA12BB@gmail.com> References: <5479127B-E572-409A-8015-C6F663CA12BB@gmail.com> Message-ID: Srigo, Thank you! It worked!! Shaks On Thu, Oct 26, 2017 at 1:53 AM, KSrigo wrote: > Hi Sharath, > > You have to do it the following way: > 1. Create the SIP profile (it will have 2 or more gateways). DO not add > gateways conf inside the SIP Profiles. > - On the top of the SIP profile add this: > > 2. Inside the folder sip_profiles create the folder gateways and add your > 2 gateways config (1 file per provider) —> Here you can define different > outboud- proxy per gateway > 3. Restart/Reload your sip_profiles > 4. From fs_cli, if you do 'sofia status gateway’, you should be able to > see your both gateways and status. > > Good luck > > Srigo > > > On Oct 25, 2017, at 7:02 PM, Sharath Kumar > wrote: > > Srigo, > > You are right. The oubound-proxy works fine too. The only problem I have > is it is global per profile. My gateways are defined in my SIP Profiles. So > what if I have 2 gateways each having different outbound-proxies :) ? > > Thank you, > Shaks > > On Wed, Oct 25, 2017 at 1:20 AM, Srigo Kanapathipillai > wrote: > >> Hi Sharath Kumar, >> >> Use the params: outbound-proxy and proxy. >> >> Eg: >> >> Outbound-proxy: sip10.voip.xyz.com >> Proxy: xyz.com >> >> Srigo >> >> Sent from my iPhone >> >> On 24 Oct 2017, at 19:04, Sharath Kumar >> wrote: >> >> Well, that is an option but I found out that the SIP Authentication >> breaks with certain providers while using the Proxy param. >> >> - The Request URI in the REG/INV will reflect the Proxy param. >> - The Proxy-Auth headers has a URI parameter that will now have the >> Request URI ( Proxy param). >> >> Some SIP providers have the Realm/Domain different from the Proxy. So >> they end up rejecting the request since the domain in the proxy is not in >> their white list of authenticated hosts. >> >> Eg. Proxy - sip10.voip.xyz.com >> Realm - xyz.com >> >> The REG/INV will have the Req URI = sip:sip10.voip.xyz.com and the Proxy >> Auth header will also have the same URI. The Providers I know are unwilling >> to add the Proxy domain sip10.voip.xyz.com into their white-list of >> authenticated hosts. They require the Proxy-Auth to have xyz.com. >> >> I am not sure if I made myself clear ? >> >> Anyway, I managed to set the sip_invite_route_uri as the Proxy. That >> seemed to work even for gateways. >> >> Thank you, >> Sharath >> >> >> On Sat, Oct 21, 2017 at 8:05 AM, Aqs Younas wrote: >> >>> Why not use proxy param in gateway? >>> >>> On 19 Oct 2017 3:11 am, "Sharath Kumar" >>> wrote: >>> >>>> Can we add an fs_path when dialing out via a gateway ? I know this has >>>> been asked before and in the archives it has been said to support and also >>>> not support fs_path in a gateway. Just wanted to confirm. >>>> >>>> thank you, >>>> Shaks. >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 mirkobrankovic at gmail.com Mon Oct 30 08:29:20 2017 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Mon, 30 Oct 2017 09:29:20 +0100 Subject: [Freeswitch-users] 180/183 not passed through sip profiles In-Reply-To: References: Message-ID: or a ringback var: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-November/101294.html On Fri, Oct 27, 2017 at 1:53 PM, Jurijs Ivolga wrote: > Hi, > > Not sure that it will help, but I think something similar was discussed > here: > > http://lists.freeswitch.org/pipermail/freeswitch-users/ > 2012-May/083985.html > > I would recommend you to set ignore_early_media inside bridge something > like this: > > > > Additionally you can try to send 180 or 183 for all calls or execute it > based on event(please check freeswitch events), never tried though, but it > may work. > > Off cause you can try to move to one profile only... > > With kind regards, > > Jurijs > > On Fri, Oct 27, 2017 at 12:39 PM, Grant Bagdasarian wrote: > >> That’s actually not true, the other FS instance uses bypass-media true. >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Grant Bagdasarian >> *Sent:* vrijdag 27 oktober 2017 11:36 >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] 180/183 not passed through sip profiles >> >> >> >> Hi Mirko, >> >> >> >> I’ll try that. >> >> >> >> To answer Alexandr Popov’s comment, we also have a FS instance running >> with just a single sip profile that also does a bridge to an endpoint on >> the internet, but uses SIP-ALG provided by our firewall, which works just >> fine. Not sure if this is related to multiple sip profiles. That’s the only >> difference between the two. >> >> >> >> Regards, >> >> >> >> Grant >> >> >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org ] *On >> Behalf Of *Mirko Brankovic >> *Sent:* vrijdag 27 oktober 2017 11:30 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] 180/183 not passed through sip profiles >> >> >> >> Hi, >> >> Have you tried with bridge_early_media=false, since in pproxy_mode >> everything that has to do with live media will fail? >> >> >> >> --mirko >> >> >> >> On Fri, Oct 27, 2017 at 9:45 AM, Grant Bagdasarian wrote: >> >> Hi, >> >> >> >> We’re using a freeswitch which bridges calls between two networks: >> private and public. The server has two interfaces. The calls originate from >> the private network and are bridged using the public sip profile. >> >> For some reason I cannot get the 180 and 183 messages passed back from >> the outbound leg to the inbound leg when sending calls over the public sip >> profile. >> >> >> >> Both sip profiles have been set with: >> >> >> >> >> >> >> >> The below dialplan is hit when a call arrives on the LAN(private) >> interface of the freeswitch. >> >> >> >> >> >> >> >> >> >> >> >> > require-nested="false"> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > /> >> >> >> >> >> >> >> >> >> >> >> >> Any ideas why the 180/183 are not relayed from outbound leg back to >> inbound leg? A trace shows these messages do arrive on the outbound leg. >> >> >> >> Grant Bagdasarian >> >> Developer >> >> +31765727054 >> >> cm.com >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> Regards, >> >> Mirko >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 4981 bytes Desc: not available URL: From telishisheer at gmail.com Mon Oct 30 09:05:14 2017 From: telishisheer at gmail.com (Shisheer Teli) Date: Mon, 30 Oct 2017 14:35:14 +0530 Subject: [Freeswitch-users] How to encrypt user extension password Message-ID: Dear Team, I am able to see user password in user extension configuration. Can anyone help to encrypt user extension password -- Regards, Shisheer T *Save paper, save trees.... Please do not print this e-mail unless it is absolutely necessary.* -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Mon Oct 30 09:19:31 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 30 Oct 2017 09:19:31 +0000 Subject: [Freeswitch-users] How to encrypt user extension password In-Reply-To: References: Message-ID: You can use a1-hash Calculator http://worman.sibhoster.ru/SipDigest.php Example https://prnt.sc/h3st1d FreeSwitch manual пн, 30 окт. 2017 г. в 12:05, Shisheer Teli : > Dear Team, > > I am able to see user password in user extension configuration. > > Can anyone help to encrypt user extension password > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > > > -- > Regards, > Shisheer T > > *Save paper, save trees.... Please do not print this e-mail unless it is > absolutely necessary.* > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/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 netcentrica at gmail.com Mon Oct 30 15:56:10 2017 From: netcentrica at gmail.com (Adam Raszynski) Date: Mon, 30 Oct 2017 16:56:10 +0100 Subject: [Freeswitch-users] Optimal trunk capacity filling algorithm Message-ID: Hi All, I'm looking for a solution to a problem that is not FreeSWITCH specific, but for sure some of you have dealt with this, and maybe someone can suggest optimal solution, so I could not re-invent the wheel It's all about proper filling of available termination trunk capacity, but not the trivial one with simple LCR Let's say we have two trunks: - Trunk 1 - capacity 100 channels, termination cost 1 cent - Trunk 2 - capacity 200 channels, termination cost 3 cent And we have clients sending wholesale traffic with random number of channels: - Client 1 - sell rate 2 cent - Client 2 - sell rate 4 cent As you see we call only sell to client 1 with profit using trunk 1. If we use LCR and client 2 has more traffic, than it will fill trunk 1 capacity and client 2 will hit trunk 2 and we will start to loose Sticking client 1 to trunk 1 and client 2 to trunk 2 is also not optimal. Number of channels is more or less random. If we stick client 2 to trunk 2 when client 1 is not using all 100 channels, than we loose additional 1 cent of margin I think about some form of capacity-aware dynamic routing - send client 2 traffic to trunk 1 only if client 1 cannot fill trunk 1, maybe using some capacity averages from last X minutes. Any suggestions how to approach this problem? Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Mon Oct 30 16:27:01 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 30 Oct 2017 16:27:01 +0000 Subject: [Freeswitch-users] Optimal trunk capacity filling algorithm In-Reply-To: References: Message-ID: Neet to implement limits for client2 on trunk1. If on trunk1 is active 80 calls, then route to trunk2 пн, 30 окт. 2017 г., 18:57 Adam Raszynski : > Hi All, > > I'm looking for a solution to a problem that is not FreeSWITCH specific, > but for sure some of you have dealt with this, and maybe someone can > suggest optimal solution, so I could not re-invent the wheel > > It's all about proper filling of available termination trunk capacity, but > not the trivial one with simple LCR > > Let's say we have two trunks: > - Trunk 1 - capacity 100 channels, termination cost 1 cent > - Trunk 2 - capacity 200 channels, termination cost 3 cent > > And we have clients sending wholesale traffic with random number of > channels: > - Client 1 - sell rate 2 cent > - Client 2 - sell rate 4 cent > > As you see we call only sell to client 1 with profit using trunk 1. If we > use LCR and client 2 has more traffic, than it will fill trunk 1 capacity > and client 2 will hit trunk 2 and we will start to loose > > Sticking client 1 to trunk 1 and client 2 to trunk 2 is also not optimal. > Number of channels is more or less random. If we stick client 2 to trunk 2 > when client 1 is not using all 100 channels, than we loose additional 1 > cent of margin > > I think about some form of capacity-aware dynamic routing - send client 2 > traffic to trunk 1 only if client 1 cannot fill trunk 1, maybe using some > capacity averages from last X minutes. > > Any suggestions how to approach this problem? > > Kind Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rw at panorgan.ch Tue Oct 31 09:35:32 2017 From: rw at panorgan.ch (=?UTF-8?Q?Ren=c3=a9_Weiss?=) Date: Tue, 31 Oct 2017 10:35:32 +0100 Subject: [Freeswitch-users] Questions about sending DTMF outband Message-ID: <441533f6-e7f7-b881-b2d4-6a78013623a9@panorgan.ch> Hi, I have some questions about sending outband DTMF tones: 1.) What is the right way to send DTMF tones to a parked call? In my tests "uuid_send_dtmf" seems to work only if I first send something (like a silent WAV file) with "uuid_broadcast". (I'm testing with Yealink SIP phones and X-Lite on my computer) Is this the right way to do it, or should I do something else to "initialize" the connection? 2.) Is there a way to detect if all DTMF tones have been played? It seems to me that "uuid_send_dtmf" returns before all tones have been played and I don't see any Freeswitch event triggered after the DTMF sequence has ended. 3.) Can I make "uuid_send_dtmf" play faster? "uuid_send_dtmf 1234 at 1" plays much slower than for example "uuid_broadcast tone_stream://w=100;d=100;1234" probably because of the "Queue digit delay of 40m" after each tone. Can I reduce this delay or is this a limitation of the DTMF outband protocol? Thank you in advance, René From udy786 at gmail.com Tue Oct 31 11:46:25 2017 From: udy786 at gmail.com (Uday kumar) Date: Tue, 31 Oct 2017 17:16:25 +0530 Subject: [Freeswitch-users] Call send from Freeswitch to Asterisk Message-ID: Hello All, I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING". DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001 *FS dialplan =>* Added DID Provider IP and Asterisk IP in ACL. When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error. *On Asterisk =>* SIP.conf [freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal EXTENSIONS.conf exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup Tried on port 5080 and 5060. Error both time same. My point is, this is Asterisk issue or Freeswitch? Please advice. -- Thanks & Regard Uday. Mobile:- +91-9377579349 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: error.PNG Type: image/png Size: 49774 bytes Desc: not available URL: From sangeeth at xoyal.com Thu Oct 26 19:24:01 2017 From: sangeeth at xoyal.com (sangeeth at xoyal.com) Date: Thu, 26 Oct 2017 14:24:01 -0500 Subject: [Freeswitch-users] call forward failing with enterprise originates Message-ID: Hi We have an extension(1000) registered at an IP phone which have a call forwarding to an external number. and the other extension(1002) on other ip phone. when i make an enterprise originate to two destinations. the 1000 gives me infinite invite requests instead of forwarding. and stop ringing. i refered to the Jira https://freeswitch.org/jira/browse/FS-6814 but looking if it had been resolved yet . BR, sangeeth kumar. From dgreenwald at gmail.com Fri Oct 27 14:03:29 2017 From: dgreenwald at gmail.com (Daniel Greenwald) Date: Fri, 27 Oct 2017 10:03:29 -0400 Subject: [Freeswitch-users] SRV and sip timers Message-ID: When the preferred server in an SRV record is down. FS waits a full 30 seconds before trying the next lower priority server. Each subsequent call also delays 30 seconds. Any tricks to speed up the failover? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mmeek at livexchange.com Mon Oct 30 20:37:26 2017 From: mmeek at livexchange.com (Matthew Meek) Date: Mon, 30 Oct 2017 20:37:26 +0000 Subject: [Freeswitch-users] SRTP scaling issues Message-ID: We have been using Freeswitch as our SBC and core switch for many years without issues. Recently we added public facing softphones using an additional server running Kamailio as TLS and webrtc bridge with FS handling media back to our core FS farm. We run a mix of webrtc and Zoiper softphones. All running SIPS (tls) but webrtc is SRTP (PCMU) and Zoiper is RTP (PCMU). We have been ramping up the webrtc traffic and at some magic point at busy times (~150 concurrent calls) the RTT and Packet loss for the webrtc calls goes crazy (as measured in the browser via RTCP and captured back on our logging servers). RTT jumps from 100ms to 2-3 seconds for most or all webrtc users and packet loss jumps from nothing to troubling on the Sending side back to FS only. Packets received are not an issue on webrtc side... webrtc can hear end caller, but end caller cannot hear webrtc well. Non webrtc traffic (Zoiper) have no reported impact. Server CPU (dual socket 6 core) looks great (no core over 20%) according to nmon... packets per second within reason (10Gb NICs). I am assuming the SRTP has some hidden scaling issue since this is the only difference between the two types of softphones. Has anyone seen this or have a workaround? I am at a loss. I assume if it is encryption overhead the CPU would be saturated and it is not. Why a single direction of SRTP traffic is impacted has me up late at night. A side note... does anyone know of a SIPP replacement that does SRTP so I can test load against this to isolate? Thanks, Matthew Meek -------------- next part -------------- An HTML attachment was scrubbed... URL: From rovshanb at yandex.ru Tue Oct 31 10:11:59 2017 From: rovshanb at yandex.ru (Rovshan Baghirov) Date: Tue, 31 Oct 2017 14:11:59 +0400 Subject: [Freeswitch-users] Prevent mod_callcenter from auto answering channel Message-ID: <190881509444719@web31j.yandex.ru> Hello, While caller waits in a queue using mod_callcenter and listens to hold music I do not want caller to be charged by operator. And only start charging when an agent picks up a call. I tried to play some announcement to caller without sending to queue and a caller is able to listen to announcement without being charged. But when I redirect user to mod_callcenter it automatically answers the session. It is coded in mod_callcenter.c line line 2795: /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ switch_channel_answer(member_channel); My FreeSWITCH version is: 1.6.19-36-7a77e0b~64bit (-36-7a77e0b 64bit) Now I want to remove this line and add it on line 3051 just before: switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(member_session), SWITCH_LOG_DEBUG, "Member %s <%s> is answered by an agent in queue %s\n", switch_str_nil(switch_channel_get_variable(member_channel, "caller_id_name")), switch_str_nil(switch_channel_get_variable(member_channel, "caller_id_number")), queue_name); I wanted to ask am I doing it right and will it have any side effects except dtmf not working during wait? Here is link to file version I'm talking about: https://freeswitch.org/stash/projects/FS/repos/freeswitch/raw/src/mod/applications/mod_callcenter/mod_callcenter.c?at=refs%2Ftags%2Fv1.6.19 Best Regards, Rovshan From del_64 at hotmail.com Tue Oct 31 11:49:06 2017 From: del_64 at hotmail.com (D-Roy Black) Date: Tue, 31 Oct 2017 11:49:06 +0000 Subject: [Freeswitch-users] Is there a way I could separate by destination types by hardcoding it in? Message-ID: Normally calls coming in from the PSTN to an unprovisioned telephone number are given session:hangup("UNALLOCATED_NUMBER"); Now I have a setup like this, that any call coming in from the PSTN to an unprovisioned telephone number is sent to an IP address, and works fine... session:execute('bridge', 'sofia/internal/gateway/'..dialledNumber..'@100.100.100.100:5061'); But is there a way I could separate by destination types by hardcoding it in? i.e if a caller dialing to 0800 or 0208 UK (sofia/internal/gateway/44800 (sofia/internal/gateway/44208 Any number beginning 44800 should go to session:hangup("UNALLOCATED_NUMBER"); And anything ells would go to the IP address session:execute('bridge', 'sofia/internal/gateway/'..dialledNumber..'@100.100.100.100:5061'); Thanks :) Deltukru From vma at vallimamod.org Tue Oct 31 15:23:07 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Tue, 31 Oct 2017 16:23:07 +0100 Subject: [Freeswitch-users] SRV and sip timers In-Reply-To: References: Message-ID: <3A9057D1-D8B5-424B-A304-411B4C150B65@vallimamod.org> Hi, You can override the default timers with timer-T1 and timer-T1X64 profile params. You can set the last one to 1000 to reduce the failover to 1s for example. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 27 Oct 2017, at 16:03, Daniel Greenwald wrote: > > When the preferred server in an SRV record is down. FS waits a full 30 seconds before trying the next lower priority server. Each subsequent call also delays 30 seconds. Any tricks to speed up the failover? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gregor at infomedia.si Tue Oct 31 17:04:52 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Tue, 31 Oct 2017 18:04:52 +0100 Subject: [Freeswitch-users] continue on fail Message-ID: Hi! Need some advice. I want to bidge call to different providers sequentialy and want to specify my own error codes where bridge would considered as failed. For this scenario is variable continue on fail. What bothers me is if I have to set variable before each bridge in dialplan or is suffice on top of dialplan: set continue_on_fail bridge bridge bridge or set continue_on_fail bridge set continue_on_fail bridge set continue_on_fail bridge ​Thank you, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Tue Oct 31 17:07:27 2017 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 31 Oct 2017 22:07:27 +0500 Subject: [Freeswitch-users] Call send from Freeswitch to Asterisk In-Reply-To: References: Message-ID: Hi, I can see type=peer, i guess that is only to put calls on the IP, you need to either use type friend or user. Regards Abbasi On Tue, Oct 31, 2017 at 4:46 PM, Uday kumar wrote: > Hello All, > > I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from > provider and I am trying to send Call to a Asterisk server but getting an > error "MANDATORY_IE_MISSING". > > DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => > Ext 8001 > > *FS dialplan =>* > > > > > > > > > Added DID Provider IP and Asterisk IP in ACL. > > When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". > Attached image is of error. > > *On Asterisk =>* > > SIP.conf > > [freeswitch_1] > type=peer > host=192.168.1.70 > port=5080 > disallow=all > allow=ulaw > trustrpid=yes > sendrpid=yes > context=from-internal > > EXTENSIONS.conf > > exten => _XXX.,1,Dial(SIP/8001,30,oT) > exten => _XXX.,n,Hangup > > Tried on port 5080 and 5060. Error both time same. > > > My point is, this is Asterisk issue or Freeswitch? Please advice. > > > -- > Thanks & Regard > Uday. > Mobile:- +91-9377579349 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From richard at treeboxsolutions.com Tue Oct 31 18:20:10 2017 From: richard at treeboxsolutions.com (Richard Chan) Date: Wed, 1 Nov 2017 02:20:10 +0800 Subject: [Freeswitch-users] SRTP scaling issues In-Reply-To: References: Message-ID: Just an idea: to isolate you use something else just to offload SRTP: e.g use rtpengine for SRTP/RTP bridging. This could show whether FreeSWITCH+SRTP is really to blame. You could even run rtpengine on the same box as FreeSWITCH since the CPU load looks manageable. Media path would look like: webrtc --[SRTP] -- (rtpengine+kamailio-to-rewrite-SDP) --[ RTP] -- FreeSWITCH On Tue, Oct 31, 2017 at 4:37 AM, Matthew Meek wrote: > We have been using Freeswitch as our SBC and core switch for many years > without issues. Recently we added public facing softphones using an > additional server running Kamailio as TLS and webrtc bridge with FS > handling media back to our core FS farm. We run a mix of webrtc and Zoiper > softphones. All running SIPS (tls) but webrtc is SRTP (PCMU) and Zoiper is > RTP (PCMU). > > > > We have been ramping up the webrtc traffic and at some magic point at busy > times (~150 concurrent calls) the RTT and Packet loss for the webrtc calls > goes crazy (as measured in the browser via RTCP and captured back on our > logging servers). RTT jumps from 100ms to 2-3 seconds for most or all > webrtc users and packet loss jumps from nothing to troubling on the Sending > side back to FS only. Packets received are not an issue on webrtc side… > webrtc can hear end caller, but end caller cannot hear webrtc well. Non > webrtc traffic (Zoiper) have no reported impact. Server CPU (dual socket 6 > core) looks great (no core over 20%) according to nmon… packets per second > within reason (10Gb NICs). > > > > I am assuming the SRTP has some hidden scaling issue since this is the > only difference between the two types of softphones. Has anyone seen this > or have a workaround? > > > > I am at a loss. I assume if it is encryption overhead the CPU would be > saturated and it is not. Why a single direction of SRTP traffic is impacted > has me up late at night. > > > > A side note… does anyone know of a SIPP replacement that does SRTP so I > can test load against this to isolate? > > > > Thanks, > > Matthew Meek > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Richard Chan Chief Architect TreeBox Solutions Pte Ltd 1 Commonwealth Lane #03-01 Singapore 149544 Tel: 6570 3725 http://www.treeboxsolutions.com Co.Reg.No. 201100585R -------------- next part -------------- An HTML attachment was scrubbed... URL: From mmeek at livexchange.com Tue Oct 31 18:42:11 2017 From: mmeek at livexchange.com (Matthew Meek) Date: Tue, 31 Oct 2017 18:42:11 +0000 Subject: [Freeswitch-users] SRTP scaling issues In-Reply-To: References: Message-ID: I already know that if not SRTP then FS has no problem with the load. I have no experience with rtpengine, so this brinngs in a new piece that I would also need to load test before sliding into production. I may just have to bring up some FS servers in the cloud and use them to generate load SRTP traffic unless someone has a better idea… From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Chan Sent: Tuesday, October 31, 2017 11:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SRTP scaling issues Just an idea: to isolate you use something else just to offload SRTP: e.g use rtpengine for SRTP/RTP bridging. This could show whether FreeSWITCH+SRTP is really to blame. You could even run rtpengine on the same box as FreeSWITCH since the CPU load looks manageable. Media path would look like: webrtc --[SRTP] -- (rtpengine+kamailio-to-rewrite-SDP) --[ RTP] -- FreeSWITCH On Tue, Oct 31, 2017 at 4:37 AM, Matthew Meek > wrote: We have been using Freeswitch as our SBC and core switch for many years without issues. Recently we added public facing softphones using an additional server running Kamailio as TLS and webrtc bridge with FS handling media back to our core FS farm. We run a mix of webrtc and Zoiper softphones. All running SIPS (tls) but webrtc is SRTP (PCMU) and Zoiper is RTP (PCMU). We have been ramping up the webrtc traffic and at some magic point at busy times (~150 concurrent calls) the RTT and Packet loss for the webrtc calls goes crazy (as measured in the browser via RTCP and captured back on our logging servers). RTT jumps from 100ms to 2-3 seconds for most or all webrtc users and packet loss jumps from nothing to troubling on the Sending side back to FS only. Packets received are not an issue on webrtc side… webrtc can hear end caller, but end caller cannot hear webrtc well. Non webrtc traffic (Zoiper) have no reported impact. Server CPU (dual socket 6 core) looks great (no core over 20%) according to nmon… packets per second within reason (10Gb NICs). I am assuming the SRTP has some hidden scaling issue since this is the only difference between the two types of softphones. Has anyone seen this or have a workaround? I am at a loss. I assume if it is encryption overhead the CPU would be saturated and it is not. Why a single direction of SRTP traffic is impacted has me up late at night. A side note… does anyone know of a SIPP replacement that does SRTP so I can test load against this to isolate? Thanks, Matthew Meek _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Richard Chan Chief Architect TreeBox Solutions Pte Ltd 1 Commonwealth Lane #03-01 Singapore 149544 Tel: 6570 3725 http://www.treeboxsolutions.com Co.Reg.No. 201100585R -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Tue Oct 31 15:46:38 2017 From: social at bohboh.info (Social Boh) Date: Tue, 31 Oct 2017 10:46:38 -0500 Subject: [Freeswitch-users] Query a Database Message-ID: Hello List, which is the best way to query a Database and use the results in Dialplan? Thank you. Regards -- --- I'm SoCIaL, MayBe From luis.daniel.lucio at gmail.com Tue Oct 31 18:04:40 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 31 Oct 2017 14:04:40 -0400 Subject: [Freeswitch-users] Double DSP payload? Message-ID: Hi Is there a reason why freeswitch sends a double DSP payload in the SIP invite from a gateway? The problem is that the payload is too big that It gets fragmented and then it fails because of that. v=0 o=FreeSWITCH 1509455513 1509455514 IN IP4 208.90.97.6 s=FreeSWITCH c=IN IP4 208.90.97.6 t=0 0 m=audio 16724 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp-mux a=rtcp:16724 IN IP4 208.90.97.6 a=ptime:30 m=audio 16724 RTP/AVP 18 0 8 3 101 13 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp-mux a=rtcp:16724 IN IP4 2 Any comments? -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: