October 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Database docs?
by Antony Stone 22 Oct '21

22 Oct '21

Hi. https://freeswitch.com/index.php/2019/08/20/freeswitch-v1-10-1-release/ tells me that there's a 2-year old module called mod_mariadb, but I cannot find anywhere on https://freeswitch.org/confluence/ the documentation about how to use it. Can someone kindly point me in the right direction? Also, just in case there's a simple answer, why isn't mod_odbc_query built into the Debian release packages available at https://files.freeswitch.org/repo/deb/debian-release/ ? I'm running FS 1.10.6 Thanks, Antony. -- "Hi, I've found a fault with the English language and I need an entomologist." "I think you mean an etymologist." "No. It's a bug, not a feature." Please reply to the list; please *don't* CC me.

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UDP/DTLS/SCTP webrtc-datachannel Freeswitch 1.10.6 and sip.js
by Jonathan Hunter 21 Oct '21

21 Oct '21

Hi guys, Just wondered if anyone had suggestions around this or had this issue before, we have 2 web clients using the sip.js library and we are trying to use the data channel to send messages to one another. If I bypass media and send media peer to peer it works fine, however if I try and send an invite with the data channel defined; m=application 52004 UDP/DTLS/SCTP webrtc-datachannel Freeswitch seems to consume this when bridging to the target end point/user. Does freeswitch support SCTP/ webrtc-datachannel if I wanted to present it from leg A to leg B so the media can flow via freeswitch? This is using debian:buster container. Thanks Jon

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RE: Recovery capabilities
by Chris B. Ware 21 Oct '21

21 Oct '21

Hi, I'm playing with, so called, FS High Availability recently. I've had the same feeling of a work in progress features.It works on resuming calls (just few seconds of audio missing during the switchover) and probably keeps the status of the call, also respect the dialplan. It is basically thought to handle two FS box with keepalived managing a shared IP, so if you need to shutdown one of the FS nodes ( I need it in my project)it doesn't work, since the dying FS will drop all the calls, cleaning Recovery table on DB. In 2014 there was a project for a module called mod_ha_cluster, I think is a dead project. Anyway this is a feature that would deserve more attention, indeed. KR,Chris ------------------------------ Message: 3Date: Mon, 18 Oct 2021 22:00:51 +0000From: "Telium Technical Support" <support(a)telium.io>To: <freeswitch-users(a)lists.freeswitch.org>Subject: [Freeswitch-users] Recovery capabilitiesMessage-ID: <0100017c956bef7e-c066e025-cdd8-4e7c-8ee4-94d02bd8ddda-000000(a)email.amazonses.com> Content-Type: text/plain; charset="utf-8" I've been reading about the recovery capabilities of FreeSwitch. It SOUNDSa bit like a work in progress (possibly waiting for funding) - but maybe thedocs are out of date. In terms of current state, can FS: 1. resume calls in progress (on restart) 2. place existing calls back into the dialplan where they where 3. restore queue agents & callers (and their order) -------------- next part --------------An HTML attachment was scrubbed...URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211018…>

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Hangup and wait for connection after dialed extension is busy.
by Paweł Femur Wojtal 21 Oct '21

21 Oct '21

Hello! Imagine the situation. I call internal extension. It is busy. So I press *0 and hangup. When leg B finishes its connection, freeswitch calls us both and bridges. How do I get such functionality? I will be grateful for any suggestion. -- fe

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playback results in silence
by Bernd Krueger-Knauber 19 Oct '21

19 Oct '21

Hi, we use mod_nibblebill and want to play a wav when the low ballance amount is reached. We noticed alreday that the example in our nibblebill.conf.xml 'play ding' is not working. There is no application play and no file ding. We use now 'playback misc/sorry.wav' for testing. We can see in the log that the file is played and we can here a silence during the playback of the file, but not the voice. If we build a small dialplan entry with 'answer', 'playback misc/sorry.wav', 'hangup', we can here the voice. Any ideas? Best regards, Bernd

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Recovery capabilities
by Telium Technical Support 18 Oct '21

18 Oct '21

I've been reading about the recovery capabilities of FreeSwitch. It SOUNDS a bit like a work in progress (possibly waiting for funding) - but maybe the docs are out of date. In terms of current state, can FS: 1. resume calls in progress (on restart) 2. place existing calls back into the dialplan where they where 3. restore queue agents & callers (and their order)

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Dial with SIP credentials without defining gateway?
by Antony Stone 15 Oct '21

15 Oct '21

Hi. I've recently started using FreeSwitch, after being familiar with Asterisk for many years. I'm wondering whether FreeSwitch has an equivalent to Asterisk's facility for dialling out to a SIP server using credentials supplied as part of the dial command, instead of needing a pre-defined gateway? In an Asterisk dialplan, you can say something like Dial(SIP/${Destination}:${SIPpass}::${SIPuser}@${SIPhost}) and that will dial to the Destination number on SIPhost using the SIP credentials SIPuser/SIPpass. Nothing needs to be configured in advance. Can FreeSwitch do something like that too? Thanks, Antony. -- "The problem with television is that the people must sit and keep their eyes glued on a screen; the average American family hasn't time for it." - New York Times, following a demonstration at the 1939 World's Fair. Please reply to the list; please *don't* CC me.

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Passing custom sip headers using re-invite
by Babak Yakhchali 13 Oct '21

13 Oct '21

Hi I need to pass some custom sip headers between 2 fs boxes after the call is established. To do this I'm using "uuid_setvar sip_h_X-myvar=value" on the channel and calling uuid_media_reneg to create a re-invite, fs is sending out the re-invite with new custom headers but the other fs box is not adding these new custom sip headers to the channel. How can I make fs to set new custom headers on the receiving side of the re-invite? fs1 ----invite---> fs2 .... call is established fs1 <---re-invite--- fs2 (re-invite with new custom sip X headers) fs1 ... not adding new headers to channel thanks

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Avoiding DDoS with verto?
by David P 13 Oct '21

13 Oct '21

After reading about recent DDoS attacks on VOIP providers in https://www.rtcsec.com/post/2021/09/massive-ddos-attacks-on-voip-providers-… in which Freeswitch is mentioned, I wondered what current practices are for services that must serve the public Internet. For example, a service that is purely verto-based seems like it could protect itself in this way: 1) Block requests on all ports (except the verto WSS login) unless the request is from an address that's already part of signaling. 2) To protect signaling, put it behind AWS API Gateway, which provides rate-limiting, and add an authorization check. Does this seem like it would defend against DDoS? Can FS be configured to do #1?

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User-to-User RFC 7433
by Adrian Worutowicz 11 Oct '21

11 Oct '21

Hello, I need to add the User-to-User field to the SIP header - for incoming calls. I have not managed to do it (neither for incoming nor for outgoing calls). In ./conf/dialplan/default.xml, <extension> section, I added : <action application="set" data="sip_h_User-to-User=123"/> or <action application="set"><![CDATA[sip_h_User-to-User=${sip_Call_ID};tenant=1101;DID=${sip_to_user}]]></action> No result. In sofia_presence.c, I've seen some code for "User-to-User", so it should be possible. But how to do it ? Best regards, Adrian.

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