- FreeSWITCH-users - FreeSWITCH Mailing Lists

forward registration
by Joli Martinez 19 Dec '20

19 Dec '20

Hello, We have a several FQDN's that are pointing to a Freeswitch box and users are registering to it. I need forward those registrations for one domain only to a new freeswitch box I am building. Not sure how to do this.

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reINVITE
by David Villasmil 17 Dec '20

17 Dec '20

Hello all, Is there any way to know which commands trigger a reINVITE? I.e.: out of these: - uuid_break - uuid_bridge - uuid_broadcast - uuid_deflect - uuid_displace - uuid_record - uuid_transfer Anyone knows off the top of their head? Thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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FreeSWITCH call recording missing header information
by Chad Phillips 16 Dec '20

16 Dec '20

I have a dialplan setup that records a bridged call in FreeSWITCH, which sets the 'api_hangup_hook' variable to call a lua script when the call hangs up, and the called lua script moves the recorded .wav file from it's temporary location to a final destination. This works, but recently I noticed that some of my .wav recordings have no header information, which results in many players choking when trying to play it. I eventually figured out that the header information for the .wav file appears to be written *after* the call is hung up, and file move operation in the api_hangup_hook moves the file before the header information has been written. This occurs sporadically. I've found that introducing a 1 second sleep in my lua script fixes this race condition, but that seems a bit of an ugly hack. Curious if anybody else has run into this, and/or can offer me a cleaner solution than my sleep hack. I'm attaching a brief summary of the dialplan and lua script for anyone that wants to see the code.

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Re: [Freeswitch-users] Copy incoming audio from participants for ASR
by David P 16 Dec '20

16 Dec '20

How do you currently forward audio to kaldi? If you use an FS conference, I think there is no way to separate the audio per-user. I agree it would be very useful if there were a way to process each user's audio separately. After a few years of using FS, I haven't come across a way to do it. --------- Forwarded message ---------- > From: R G <mitachundkrach(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 14 Dec 2020 12:40:38 +0100 > Subject: [Freeswitch-users] Copy incoming audio from participants for ASR > Hello everyone, > > i have a technical question about freeswitch and would like to know if > my idea is technically possible. > > I've got a project at the university and my goal is to improve automatic > speech recognition for bigbluebutton. > > In the summer semester a group of students of my university made a > subtitling plugin[1] for bigbluebutton. This plugin adds a participant > into the meeting and grabs his audiostream to send it to the kaldi ASR > Software. When more than one person speaks at a time the detection > decreases. > > My idea is to copy the incoming freeswitch audiostreams from every > participant and send them directly as a copy to the ASR (kaldi can > handle multiple audiostreams at once). > > In the Confluence-Wiki i searched for something in this direction but > only found things like Dialplan to ring multiple telephones at once. > > I hope my idea is formulated understandably. > > > Kind regards, > > Robert > > [1] https://github.com/3wille/bbb-kaldi-connector > >

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Full Trickle ICE via mod_sofia supported?
by Alexander Haugg 16 Dec '20

16 Dec '20

Hello, Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? If so, which settings or sip headers are required? Thanks Alex

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disable NOTIFY message
by Aidar Kamalov 16 Dec '20

16 Dec '20

Hello, is it possible to disable NOTIFY messages from freeswitch to external sip profile? NOTIFY sip:aaaa@aaaaaaa;intercom=true SIP/2.0 .... Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: talk Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Length: 0 -- Aydar A. Kamalov

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Can't test for empty variable
by David Villasmil 16 Dec '20

16 Dec '20

Hello all, I'm trying to test the play_and_get_digits variable from here: <action application="play_and_get_digits" data="1 1 1 3000 # '' silence_stream://250 dtmf_get \d 3000"/> Like this: <condition field="${dtmf_get}" expression="^$"> But it's always false... How can I test for an empty (null?) variable? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Copy incoming audio from participants for ASR
by R G 15 Dec '20

15 Dec '20

Hello everyone, i have a technical question about freeswitch and would like to know if my idea is technically possible. I've got a project at the university and my goal is to improve automatic speech recognition for bigbluebutton. In the summer semester a group of students of my university made a subtitling plugin[1] for bigbluebutton. This plugin adds a participant into the meeting and grabs his audiostream to send it to the kaldi ASR Software. When more than one person speaks at a time the detection decreases. My idea is to copy the incoming freeswitch audiostreams from every participant and send them directly as a copy to the ASR (kaldi can handle multiple audiostreams at once). In the Confluence-Wiki i searched for something in this direction but only found things like Dialplan to ring multiple telephones at once. I hope my idea is formulated understandably. Kind regards, Robert [1] https://github.com/3wille/bbb-kaldi-connector

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NORMAL_TEMPORARY_FAILURE Suddenly
by Don Hawkins 14 Dec '20

14 Dec '20

Hi everyone! Suddenly all outbound calls (ALL of our providers) are returning NORMAL_TEMPORARY_FAILURE on our FS server. I've been working on this for roughly four hours so I wanted to reach out. It does seem that at least with one provider the calls are getting to their servers since they show as "Failed" in their management portal. We've changed nothing, suddenly origination just stopped working. sofia/gateway/*Provider*/12145551212%20&park() Any input is appreciated, I know it's something simple, just need to be guided in the right direction. Thanks! Don

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Re: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE Suddenly
by Don Hawkins 14 Dec '20

14 Dec '20

Kaiduan, That's the termination provider's IP address.

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