August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto - Chrome - MediaStream 'ended' event removed
by Gregor Nanger 18 Aug '16

18 Aug '16

I get this warning in chrome when using verto: The MediaStream 'ended' event is deprecated and will be removed in M54, around October 2016. See https://www.chromestatus.com/features/5730404371791872 for more details. Can someone please elaborate what does this mean for verto js functionality?

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Re: [Freeswitch-users] Issue connecting to FS with SRTP
by Oleg Stolyar 18 Aug '16

18 Aug '16

I think you are right Sergey. Should it be fixed on the FS side or on the client SDK side though? Which is the right value? By the way, the SDK can do either dtls or sdes. This issue happens with dtls. Sdes works fine and I can establish a call. On Aug 16, 2016 10:14 PM, "Sergey Safarov" <s.safarov(a)gmail.com> wrote: Think issue related to "m=audio 61434 UDP/TLS/RTP/SAVP" Try "m=audio 61434 RTP/SAVP" ср, 17 авг. 2016 г. в 5:35, Oleg Stolyar <olegstolyar(a)gmail.com>: > Thanks Gonzalo, > > That setup on the FS side refers to TLS and as I said TLS is working for > me, I have it set up. > > SRTP should work out of the box AFAIK. I suspect there is something in > the SDP I posted that FS considers to be non-compliant with the specs. Not > sure what though and whether it's a bug in FS or in the client SDK I am > using. > > On Tue, Aug 16, 2016 at 7:22 PM, Gonzalo Gasca Meza < > gascagonzalo(a)gmail.com> wrote: > >> Did you enable/configure SRTP? This >> <http://twilio-marketing-prod.elasticbeanstalk.com/resources/images/docs/Twi…> >> is a sample guide I use for Twilio TLS/SRTP which can help >> Can you post all logs? >> >> >> On Tue, Aug 16, 2016 at 6:10 PM, Oleg Stolyar <olegstolyar(a)gmail.com> >> wrote: >> >>> Hi guys, >>> >>> I am trying to connect to FS using the CounterPath SDK for Android. I >>> use TLS for signalling. All works well until I try to enable SRTP. Then >>> FS rejects the call the INCOMPATIBLE_DESTINATION message. Is there >>> something wrong with the SDP? Here are the relevant SDPs from the INVITES. >>> >>> NON-SRTP (works) >>> v=0 >>> o=- 622419737965 1 IN IP4 192.168.1.78 >>> s= >>> c=IN IP4 192.168.1.78 >>> t=0 0 >>> m=audio 57194 RTP/AVP 3 120 123 122 121 9 8 0 84 85 18 101 >>> a=rtpmap:120 opus/48000/2 >>> a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 >>> a=rtpmap:123 SILK/24000 >>> a=rtpmap:122 SILK/16000 >>> a=rtpmap:121 SILK/8000 >>> a=rtpmap:84 speex/16000 >>> a=rtpmap:85 speex/8000 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=sendrecv >>> >>> SRTP (does not work) >>> v=0 >>> o=- 622187157154 1 IN IP4 192.168.1.78 >>> s= >>> c=IN IP4 192.168.1.78 >>> t=0 0 >>> a=fingerprint:SHA-256 4D:08:D6:49:9E:CA:77:A2:77:74: >>> 02:A0:B9:92:32:2F:2D:76:7D:59:7F:A4:CA:85:16:BA:D0:27:0A:74:1C:0F >>> a=setup:actpass >>> m=audio 61434 UDP/TLS/RTP/SAVP 3 120 123 122 121 9 8 0 84 85 18 101 >>> a=rtpmap:120 opus/48000/2 >>> a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 >>> a=rtpmap:123 SILK/24000 >>> a=rtpmap:122 SILK/16000 >>> a=rtpmap:121 SILK/8000 >>> a=rtpmap:84 speex/16000 >>> a=rtpmap:85 speex/8000 >>> a=rtpmap:18 G729/8000 >>> a=fmtp:18 annexb=yes >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-15 >>> a=sendrecv >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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INVITE timeout
by huseyin kalyoncu 17 Aug '16

17 Aug '16

Hello, When using simple bridge application in a dialplan I am using leg_timeout which is default to 60 secs. If I set call_timeout Freeswitch sends 480 after exactly 120 secs. How can I increase this value? Is there any parameter for this? Or is this value hardcoded in the source code? If I have to change the source code to increase this value, which source code do I have to modify? I foud the following line in the sofia lib tree, NHP_SET(nhp, invite_timeout, 120); but I am not sure this is the correct parameter or not. Thanks, Huseyin

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Getting all sip messages
by David Villasmil 17 Aug '16

17 Aug '16

Hello guys, I'm writing a C module, is there an EVENT i can bind to, to receive ALL SIP messages? Regards, David ᐧ

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VC no ringtone when remote ringing
by Bipin Patel 17 Aug '16

17 Aug '16

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Issue connecting to FS with SRTP
by Oleg Stolyar 17 Aug '16

17 Aug '16

Hi guys, I am trying to connect to FS using the CounterPath SDK for Android. I use TLS for signalling. All works well until I try to enable SRTP. Then FS rejects the call the INCOMPATIBLE_DESTINATION message. Is there something wrong with the SDP? Here are the relevant SDPs from the INVITES. NON-SRTP (works) v=0 o=- 622419737965 1 IN IP4 192.168.1.78 s= c=IN IP4 192.168.1.78 t=0 0 m=audio 57194 RTP/AVP 3 120 123 122 121 9 8 0 84 85 18 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:123 SILK/24000 a=rtpmap:122 SILK/16000 a=rtpmap:121 SILK/8000 a=rtpmap:84 speex/16000 a=rtpmap:85 speex/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SRTP (does not work) v=0 o=- 622187157154 1 IN IP4 192.168.1.78 s= c=IN IP4 192.168.1.78 t=0 0 a=fingerprint:SHA-256 4D:08:D6:49:9E:CA:77:A2:77:74:02:A0:B9:92:32:2F:2D:76:7D:59:7F:A4:CA:85:16:BA:D0:27:0A:74:1C:0F a=setup:actpass m=audio 61434 UDP/TLS/RTP/SAVP 3 120 123 122 121 9 8 0 84 85 18 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000 a=rtpmap:123 SILK/24000 a=rtpmap:122 SILK/16000 a=rtpmap:121 SILK/8000 a=rtpmap:84 speex/16000 a=rtpmap:85 speex/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv

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Newbie Logging question
by Shawn Wheeler 16 Aug '16

16 Aug '16

Hello, I see in the log file where calls from extension to extension are initiated. Is there a quick and dirty syntax I can look for when looking for a call from one to another. Example extension 9000 call extension 4008 Thank you in advance Shawn Shawn Wheeler Chief Evangelist shawn.wheeler(a)interlockconcepts.com<mailto:shawn.wheeler@interlockconcepts.com> Cell:602.466.8855 [interlock concepts horizontal 600 x 205]<http://www.interlockconcecpts.com/>

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video layout position event
by Mirko Brankovic 16 Aug '16

16 Aug '16

Hi, I have started this feature request: https://freeswitch.org/jira/browse/FS-9429 So trying to get tat lease log for starters, then event can come latter. If anyone have any pointers on there would be a good place to pick it up i would appriciate any help. Currently I have being trying out conference_video.c probably in conference_video_scale_and_patch should be the position of the video based on the border scale. Problem is that this is a loop that will emit a lot of lines for each channel. Smart way would be to emit on conference start, member join, and member leave. the possition of that one or maybe all members since they change in layout. Also it happens that members are changing place when talking so that will make it more difficult I guess. Thanks in advance, MIrko

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Re: [Freeswitch-users] Web sockets and gRPC modules?
by Leonardo Lima Ribeiro 16 Aug '16

16 Aug '16

Michael, thank you for the response! Actually I was thinking about copy the function “recordFile” but instead of save a file in the system etc, just the streams to my STT service… Example of usage in LUA: text = session:asrApi(“IBM_WATSON”, max_len_secs, silent_threshold, silence_secs, keywords) Which “IBM_WATSON” points me to a XML conf file (eg. conf/asrapi/IBM_WATSON.xml) which have all the settings needed to stream to that API like auth token, endpoints urls, supported protocols (initially web sockets and gRPC) etc. max_len_secs, silent_threshold, silence_secs - exactly like the current recordFile function params. keywords - optional field, some of these apis can use a list of words to improve the accuracy and speed of recognition... For now, if my approach is correct, I found the following functions, to have an idea on how to start: switch_ivr_play_say.c - switch_ivr_record_file switch_cpp.cpp - CoreSession::recordFile So yeah, I've setup already a development environment for me, following the instructions on (Creating New Modules - Confluence page). To be honest it’s my first time developing on C, so if you have good sources to improve my learning curve on “how to create a module in FreeSWITCH” would be awesome! Can you give me a feedback on my approach and also the directions/instructions in what modules/functions should I use to do that? Thank you, Leo > On Aug 15, 2016, at 11:33 AM, freeswitch-users-request(a)lists.freeswitch.org wrote: > > we would welcome for review someone submitting an module to support these.. it should not require a new module format, we already have those for speech recognition.. it would require a module that implemented the interfaces. > > >> On Aug 14, 2016, at 5:47 AM, Leonardo Lima Ribeiro <llribeiro90(a)gmail.com <mailto:llribeiro90@gmail.com>> wrote: >> >> Hello guys! >> >> Is there any modules that we can use to make ASR using IBM Bluemix Watson Speech To Text (which accepts web sockets for streaming recognition) or Google Speech to Text API (which uses gRPC for streaming recognition)? >> >> If there are none, do you think it’s worth it to build them (like in a new module format)? I know the current standard way is to use unimrcp, but unfortunately I could not get good results for my business cases using Sphinx, and the good services appears to be pricey… Said that, I was thinking about build modules that allows us to easily integrate to Google STT and IBM Watson STT Apis, which are the most popular and accessible services... >> >> Thank you, >

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append header when LOSE_RACE
by Anonim Stefan 16 Aug '16

16 Aug '16

Hi, I am facing a bridge parallel forking scenario where I need to append a header in the CANCEL send to the LOSE_RACE leg. Is there a way I can do this? Thanks, Stefan

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