March 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Installing Freeswitch
by Alexander Perkins 30 Mar '22

30 Mar '22

Hi All. I am trying to install Freeswitch on Debian 11, but I am running into some issues. Here's what I am doing: 1. Created a key in SignalWire. 2. Ran the following command (pat_XXXXX is a replacement for my actual pat): - wget --http-user=signalwire --http-password=pat_XXXXX -O /usr/share/keyrings/signalwire-freeswitch-repo.gpg https://freeswitch.signalwire.com/repo/deb/debian-release/signalwire-freesw… 3. However, this always returns 401 Unauthorized: - --2022-03-29 22:26:18-- https://freeswitch.signalwire.com/repo/deb/debian-release/signalwire-freesw… Resolving freeswitch.signalwire.com (freeswitch.signalwire.com) 2803:d000:fffe::174, 190.102.98.174 Connecting to freeswitch.signalwire.com (freeswitch.signalwire.com)|2803:d000:fffe::174|:443... connected. HTTP request sent, awaiting response... 401 Unauthorized Authentication selected: Basic realm="SignalWire Auth" Reusing existing connection to [freeswitch.signalwire.com]:443. HTTP request sent, awaiting response... 200 OK What am I doing wrong? Thank you, Alex

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Equipment supporting V.150
by Steve Underwood 29 Mar '22

29 Mar '22

Hi, This is VoIP rather than a specific Freeswitch question. I am developing an implementation of some of the V.150 series specifications within the spandsp library, to be able to support modem over IP in open source software like Freeswitch. I am having problems identifying equipment I can test against. There is quite a bit of used VoIP kit around, at E-Bay and similar places. I assume I can find something used from Cisco and others that is not too expensive, for my development testing. My problem is identifying exactly which things offer V.150 support. Looking at the Cisco documentation, they seem to focus far more on the software requirements for the support of specific features, than on the model numbers of hardware with the relevant support. Can anyone help me with this? Regards, Steve

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codec offer order
by David Villasmil 29 Mar '22

29 Mar '22

Hello all, In the vars default codecs i have OPUS,G722,PCMU,PCMA... But when i receive a call with say: PCMU+G729 FS always sends out PCMU+OPUS. I want OPUS to be first, what am I missing= Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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SRTP for Internal Calls only
by Rasheed Kalapurackal 29 Mar '22

29 Mar '22

Dear All , I have enabled TLS and SRTP for internal profile meant only for internal calls. In Vars.xml , i made rtp_secure_media = true. Also in directory default.xml , i modified the following param. <param name="dial-string" value="{rtp_secure_media=${regex(${sofia_contact(${dialed_user}@ ${dialed_domain})}|(transport=(?i)TLS))},presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> Internal calls are working well with these changes and internal calls between 2 end points are established using TLS / SRTP . For outbound gateway based calls , we opted to make it without TLS / SRTP by disabling tls <param name="tls" value="$${external_ssl_enable}"/> where external_ssl_enable is false. The late negotiation is true. It was noticed that any outbound calls made through the external gateway is also going out with RTP/SAVP in SDP and hence our ITSP is rejecting the calls with error 488 "Not acceptable here" Is there any way to resolve this issue. Any help in this regard is highly appreciated. Thanks and Regards Abdul Rasheed

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Alternatives to Loopback
by Support from NetworkedAudio LLC 26 Mar '22

26 Mar '22

I'm trying to avoid using loopback in my dialplan - I'm getting calls which are hard to clear and track. I'm doing conference dial loopback/99997 where 99997 in the default dialplan joins a second conference. Has anyone had any success with alternatives to loopback? I want to listen to one conference from another.

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Re: [Freeswitch-users] Problem with repository server?
by Mohammad-Reza Tazari 23 Mar '22

23 Mar '22

Yes, however on Raspberry Pis we cannot switch to Bullseye yet, as can be seen here: https://freeswitch.org/confluence/display/FREESWITCH/Raspberry+Pi -- Saied On Wed, 16 Mar 2022 11:19:25 -0400 botelist(a)gmail.com wrote: > > I believe you are following outdated instructions. > > Debian Bullseye is the currently supported distribution, Buster has been deprecated. > > https://freeswitch.org/confluence/display/FREESWITCH/Linux

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Concern about repository server and conditions
by Laura Smith 22 Mar '22

22 Mar '22

Hi Like many on here, I'm unconvinced by the sudden necessity to build a wall behind the repository server. In particular I don't buy the argument that it fights against third-party tampering, since surely that is what GPG signing of the repo is for ? But I digress. I have a *MAJOR* problem with its implementation. Namely I am told to go to "https://id.signalwire.com/signup/account/new" and sign up for an account. Fine, except that action has an "By continuing you agree to our Legal Policies". This is problematic for two reasons: 1) Many people, by virtue of their employment conditions, are not in a position to enter into legally binding contracts on behalf of their employer. 2) But, perhaps MOST IMPORTANTLY is that these "Legal Policies" include Section 9 "LICENSES AND PROPRIETARY RIGHT". A brief read of that licensing section seems to go firmly against the grain of OSS with its many prohibitions. I don't like it. I don't like the direction of travel that this implies. I think this might be the beginning of me pulling Freeswitch from my infrastructure, and the end of me recommending it to others. Laura

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Ansible playbook automates Debian token config
by Bote Man 21 Mar '22

21 Mar '22

Gavin Henry has kindly added an Ansible playbook recipe that automates the updating of Apt config files with your personal access token. This can be a boon if you have a lot of machines to update in a short period of time. The Confluence wiki page is at: https://freeswitch.org/confluence/display/FREESWITCH/Ansible+playbook+to+upd ate+your+Debian+repository+to+add+a+SignalWire+Token John Boteler Bote Communications

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FreeSWITCH Packages Repo. UPDATE
by Len Graham 21 Mar '22

21 Mar '22

Hello Everyone! We wanted to give a status update on the status of our FreeSWITCH Packages Repo. It's undergoing maintenance at the moment and we ran into unexpected downtime. We are working to resolve this as quickly as possible and thank you for your patience. If you have an immediate need to install FreeSWITCH this can be compiled at https://github.com/signalwire/freeswitch.

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Sending announcements to all callers
by Support from NetworkedAudio LLC 21 Mar '22

21 Mar '22

I have a system where a user dials in, get authenticated and placed into a standard conference. Other local resources get added to the conferences as needed, and that is working well. For safety reasons, we need to play a single incoming PortAudio channel to all dialed in users. If I do an auto call or originate I can add a deaf member to each conference which is a loopback to an eavesdrop. This is messy but works. However, it uses a lot of resources and now my local people have the announcements twice, with delay. What I want is just to send audio out to those dialing in. I have tried sending eavesdropping just to the incoming call but it won’t let me then join a conference through the dialplan - processing stops at the eavesdrop. I see broadcast but can only send tones or saved files. Ideally the incoming external callers would hear the announcements and not the conferences themselves. I’d like to avoid loopback, too. Does anyone have ideas, please?

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