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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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NOTIFY sent only about 1 out of 10 times when using IPV6
by Mario G 16 Aug '16

16 Aug '16

No problem with IPV4 for years. Changed Yealink and Bria phones to IPV6, they still work fine including getting VM. But with IPV6 FS only sends a NOTIFY 1 out of ten times so the phones don't set MWI. A sip trace during registration shows everything identical between MWI and non-MWI working except that when MWI does work a NOTIFY is sent to the phone. The NOTIFY, and only the NOTIFY, references IPV4 addresses in “From, To, Message-Account” and I wonder if that may be the cause. I tried to change Message-Account to IPV6 but it gets truncated at the first colon, I opened jira FS-9428 <https://freeswitch.org/jira/browse/FS-9428> for that. I am wondering if this problem is one for Jira too or a config issue? Anyone run into this or have a suggestion? Thanks. Mario G

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callee_id_number for FreeTDM channels
by Trever L. Adams 15 Aug '16

15 Aug '16

I am having difficulty getting something to work. I have FXO, FXS and sip (internal only) channels. FXS->sip has destination_number set to the number id setup in the directory. FXS->FXO sets the destination to the span number (which in this case is part of the number, but not complete). I have the name and number set properly in freetdm.conf which works for outgoing, but not incoming. I have tried setting effective_callee_id_number and origination_callee_id_number before the transfer, neither of these changes the last_sent_callee_id_number. Is there a way to either get the last_sent_callee_id_number to be correct or to get the destination number to be correct? Thank you, Trever

1 0

Chrome v52
by Mirko Brankovic 15 Aug '16

15 Aug '16

Hi, Has anyone encountered this new problem with new version of Chrome. Maybe it is a question for the Chrome dev but just to ask :) Chrome v52 gives me bad AV handshake on FS: 2016-07-27 08:52:20.952477 [ERR] switch_rtp.c:3112 video Handshake failure 1 2016-07-27 08:52:20.952477 [INFO] switch_rtp.c:3113 Changing video DTLS state from HANDSHAKE to FAIL 2016-07-27 08:52:20.982513 [ERR] switch_rtp.c:3112 audio Handshake failure 1 2016-07-27 08:52:20.982513 [INFO] switch_rtp.c:3113 Changing audio DTLS state from HANDSHAKE to FAIL On v51 fingerprint is verified: 2016-07-27 09:03:02.072474 [INFO] switch_rtp.c:3119 Changing video DTLS state from HANDSHAKE to SETUP 2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3028 video Fingerprint Verified. 2016-07-27 09:03:02.122569 [INFO] switch_rtp.c:3644 Activating Video Secure RTP SEND -- Regards, Mirko

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VertoJS audio Handshake failure 1
by Murugan Pandian 15 Aug '16

15 Aug '16

When i make call from firefox i am getting Error : audio Handshake failure 1 FS Version : 1.6.9 Firefox Version: 48

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freeswitch path layout issue in 1.6
by Caragea Silviu 15 Aug '16

15 Aug '16

Hello, >From what I see in 1.6 (I've used 1.4 since now) the layout path changed (I like the new one :) ). For example the logs are in : logfiledir: /var/log/freeswitch This is working nice but in case you specify when you start FS -base = /usr/local/something then all other path's are not respecting any longer the layout and for example the logs are in : /usr/local/something/log instead /usr/local/something//var/log/freeswitch Basically is switching to the old layout from FS 1.4 is this expected or it's a bug ? Silviu

2 3

Web sockets and gRPC modules?
by Leonardo Lima Ribeiro 15 Aug '16

15 Aug '16

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,

2 1

CentOS users - beware of the last kernel update
by Yehavi Bourvine 15 Aug '16

15 Aug '16

Hi, About a week and a half ago there was an update to CentOS 7 kernel which upgraded it from 3.10.0-327.22.2.el7 to 3.10.0-327.28.2.el7; after that upgrade we observed a few times that Freeswitch went hung - it did not respond to SIP requests, but continued to respond to ESL messages; we had to restart it, and it returned a few hours later. After two such days we returned back to the previous kernel version and now we are running without any glitch. Regards, __Yehavi:

1 0

absolute_codec_string after unsuccessful bridge
by Jurijs Ivolga 15 Aug '16

15 Aug '16

Hi! I have following use-case: My UAC makes video call(it offers VP8 and Opus codecs) and after unsuccessful bridge Freeswitch will answer a call and execute record app. When freeswitch fires answer app it offers SDP containing VP8 and OPUS, but I would like that it offers only Opus. Because freeswitch offers VP8, my UAC starts video call and record app do not work as expected. I'm getting following error: "[ERR] switch_ivr_play_say.c:496 Unable to establish inbound video stream" I need somehow to set absolute_codec_string after unsuccessful bridge. I tried but it didn't worked. If freeswitch answers immediately without bridge, then I'm able to set absolute_codec_string. Is there any possibility to set absolute_codec_string after unsuccessful bridge? Or maybe there is other possibilities to record audio message during video call? Jurijs

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NAT/VPN DTMF and Audio Issue
by Kieran Hackshall 15 Aug '16

15 Aug '16

We have a Freeswitch instance setup and it is working for calls using SIP to carriers. The current solution we have uses a NAT server to present a single IP to carriers. The issue we are facing now is a new carrier that required us to connect via VPN and then handle RTP over the internet via a NAT. Currently we have setup the new requirement and can make outbound calls and play and hear media, however, DTMF is not working. Also when a call is transferred no audio is present on either side. One major difference I can see between using our current carriers that are woking and the new one is that I cannot see in the Freeswitch log 'RTP Correct ip/port confirmed'. Not too sure if this indicates there is an issue or not. Our Freeswitch configs are vanilla except that we set the NAT external IP for SIP and RTP in the vars.xml I have attached a Freeswitch log with additional SIP tracing, any help in resolving this issue would be greatly appreciated. Thanks, Kieran

1 0