September 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with the "file-only" Conference Layout Option
by Lyle Pratt 15 Sep '20

15 Sep '20

Howdy Folks, I'm experimenting with some video capabilities and am trying to play a video file in a specific layout position in a conference. From the documentation, it seemed like I could use the "file-only" option in a layout configuration to achieve this, however, even with the layout applied on the conference, video files always play full screen. >From the docs: file-only with file-only="true" this box will only show video files that are played via the conference play api. Does the "file-only" option no longer work? Any other tips? Thanks! Lyle Pratt

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[mod_amqp][golang]
by Bilal Abbasi 15 Sep '20

15 Sep '20

Hi Users, I am using mod_rabbitmq to send events on rabbitmq and trying to capture/consume using golang. I saw that there are certain things we need from mod_amqp, like queue name,binding key, consumer tag. I cant see them under amqp.conf and they are required by golang amqp script https://github.com/streadway/amqp /_examples/simple-consumer/consumer.go Need help Regards Abbasi

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issue with mod_python
by David Villasmil 12 Sep '20

12 Sep '20

Hello guys, i¡m not sure what's going on. I just installed a fresh 1.10.5 and i've got a script in # ls -l /usr/share/freeswitch/scripts/foo-outbound.py -rw-r--r-- 1 root root 3804 Sep 8 23:44 /usr/share/freeswitch/scripts/foo-outbound.py but FS either can't find it or something's changed in the architecture: 2020-09-09 00:05:08.548244 [NOTICE] mod_python.c:213 Invoking py module: foo-outbound 2020-09-09 00:05:08.548244 [ERR] mod_python.c:253 Error importing module 2020-09-09 00:05:08.548244 [ERR] mod_python.c:165 Python Error by calling script "foo-outbound": <type 'exceptions.ImportError'> Message: No module named foo-outbound l> status UP 0 years, 0 days, 0 hours, 26 minutes, 27 seconds, 2 milliseconds, 76 microseconds FreeSWITCH (Version 1.10.5 -release-17-25569c1631 64bit) is ready 6 session(s) since startup 0 session(s) - peak 1, last 5min 0 0 session(s) per Sec out of max 30, peak 1, last 5min 0 1000 session(s) max min idle cpu 0.00/90.07 Current Stack Size/Max 240K/8192K Anyone knows anything about this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Early media in the "wrong" direction
by Olle Frimanson 11 Sep '20

11 Sep '20

Hi, We have a use case where we need to send early media from A-leg and play it before 200 OK on multiple B-legs. This the scenario in principle: A sends an INVITE to Freeswitch Freeswitch responds with 183 A starts sending media to Freeswitch Freeswitch does parallell bridging to B,C and D B,C and D responds with 183 Now I want Freeswitch to relay the media transmitted from A to B,C and D Is this scenario supported by Freeswitch I have various options like audio_media_flow and bridge_early_media but I can't get it to work. Any help or hints are appreciated BR/Olle

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[In Progress/Ringing Calls][FreeSWITCH Event]
by Bilal Abbasi 10 Sep '20

10 Sep '20

Hi Users, I am trying to get in progress call event from freeswitch, i saw that we have channel_progress and channel_progress_media events, but those are not getting executed all the time. Can someone guide me what event i need to capture to show current running calls, basically it would be a simple counter getting increment on new channel in progress/ringing and decrement on channel_hangup_complete event. Regards Abbasi

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How to avoid clipping when transcoding to PCMA/U ?
by David P 10 Sep '20

10 Sep '20

When we configure FSv10.4 to transcode Opus to PCMA/U, the output stream always has clipping (peak amplitude is always -0.17dB, 0.99 linear when measured in Audacity). I tried configuring opus.conf.xml as suggested in https://serverfault.com/a/807070 (with restart via: fs_cli -x 'fsctl shutdown elegant restart') but it makes no difference. When we simultaneously record the session with Audacity clientside, less than half the available range is used, so the user mic doesn't have gain too high. How can we avoid clipping when transcoding?

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 171, Issue 7
by Tazari, Mohammad Reza 10 Sep '20

10 Sep '20

Tom, first: I hope that you noticed that in my latest email, I reported that I am still not able to make use of mod_portaudio; so the instructions I provided on Aug 26 helped to have some progress on my side, but they have not been sufficient... Now, you said: you "just need to be in the proper group"; I would just add the importance of the proper user to this statement, as well. The page https://freeswitch.org/confluence/display/FREESWITCH/Debian+Post-Install+Ta… suggests a systemd unit file for FS that includes the directive "User=root". But, there is also the comment saying that FS starts "as root, so Freeswitch can set its priority, create some directories if needed, etc. Then it _*will drop privileges to continue running as user and group Freeswitch*_." Are you using that same unit file? If so, then the following part of my instructions will be essential for the success: *sudo adduser freeswitch audio sudo adduser ***freeswitch *bluetooth* Otherwise, you must find out as which user the FS is eventually running in your system and then run the above two commands for that user. I gave up using the jack because it is only for output, but in my application I need both input and output. In the beginning that I didn't know this and was hoping to use the jack for my application, I used the amixer command few times, but I am not very familiar with this command, nor with alsamixer. From what I have understood so far: * aplay -l and arecord -l will show only your audio devices on sound cards of type hardware (I think that /usr/share/alsa/alsa.conf is somehow involved in this) * aplay -L and arecord -L will also show additional devices defined in /etc/asound.conf and the current user's ~/.asoundrc (such as virtual devices and real devices on virtual sound cards) * mod_portaudio has been only able to see my real devices via bluealsa which acts as a kind of virtual sound card, nothing else!!! This is very strange to me and I haven't found any explanation for that.... Kind regards, -- Saied freeswitch-users-request(a)lists.freeswitch.org wrote on 07-Sep-20 18:07: > > ForwardedMessage.eml > > Subject: > Re: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? > From: > Tom Lynn <tom(a)tomlynn.com> > Date: > 07-Sep-20, 18:06 > > To: > FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Saied, > You are a very determined man! I followed your instructions and yes, > this works when I run it from the command line as root. Again, when > launched by systemd I come up with no reply when I run pa devlist in > fs_cli. > > When free time is more plentiful, I will investigate trying to run > this as a different user. I think I just need to be in the proper > group. I will also need to adapt this to use the 3.5mm output of the > pi. Where do you see the device names, in the alsa mixer?

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Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP
by Shaun Stokes 09 Sep '20

09 Sep '20

Hi All, We are using '3pcc-enable' to allow late codec negotiation where an INVITE (or RE-INVITE) does not include an SDP but have recently noticed unexpected behaviour on certain calls with Broadsoft equipment. The problem is on outbound calls from FreeSWITCH to BT IP Exchange which are handled by a 3rd Party using Cisco (previously Broadsoft) equipment, calls to the 3rd party that are routed to an IVR then to a hunt group (ACD) have the 'a=sendonly' attribute set in the SDP by the 3rd party (Broadsoft) which they expect us to remove in our 2xx response to a RE-INVITE with-out an SDP (from 3rd party) according to 'RFC 3261 section 14.2' as we "SHOULD construct the offer as if the UAS were making a brand new call" and "this means that it SHOULD include as many media formats and media types that the UA is willing to support", I've read into this further and find the interpretation to be somewhat ambiguous as there is a reference to 'RFC 3264 section 8' when an offer "updates an existing session" then "the offer MAY be identical to the last SDP" so technically both arguments are correct but unfortunately BT is siding with Broadsoft at this stage which is used by a variety of large service providers whom all agree on this interpretation. The 3rd party have also stated that this isn't a call going on hold as it's routing to an ACD group, according to 'RFC 6337 section 5.3' "the use of sendonly/recvonly is not limited to hold". It's not very well documented but I suspect setting 'sip_unhold_nosdp' in FreeSWITCH may resolve the problem as a workaround but this requires further testing. Should the default behaviour of FreeSWITCH be changed when '3pcc-enable' is used in this situation so FreeSWITCH creates a brand new SDP or are Broadsoft wrong and does 'RFC 3261 section 14.2' need to be updated and if so how? Thanks, Shaun

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Re: [Freeswitch-users] Handle media of all calls via port 443?
by David P 09 Sep '20

09 Sep '20

I did: fs_cli -x 'fsctl shutdown elegant restart' but that still yields only the same candidate as before. Then I rebooted the vm (ec2), tried again, and there was still just one candidate of the same type as before. On Wed, Aug 12, 2020 at 5:18 AM < freeswitch-users-request(a)lists.freeswitch.org> wrote: > ---------- Forwarded message ---------- > From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> > > In this case, you *must* restart freeswitch, reloadxml is not enough > -giovanni > > On Tue, Aug 11, 2020 at 9:01 PM David P <davidswalkabout(a)gmail.com> wrote: > >> Thank you for your suggestion, Sergey. However, I found that introducing >> those two settings (in FSv10.4 on Debian10) made no difference; I still got >> just one candidate in FS's answer SDP and it's udp rather than tcp (for >> port 443)... >> >> Before introducing >> <X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/> in >> /etc/freeswitch/vars.xml >> and >> <action application="set" data="media_webrtc=true"/> in >> /etc/freeswitch/dialplan/mydialplan.xml >> the browser receives Answer SDP containing only one ice candidate: >> a=candidate:5500513041 1 udp 659136 35.xxx.yy.zzz 26928 typ host >> generation 0 >> >> After introducing those two config changes, and `fs_cli -x "reloadxml"`, >> the set of candidates in the answer the browser gets is the same: >> a=candidate:4793838179 1 udp 659136 35.xxx.yy.zzz 19518 typ host >> generation 0 >> >> The reason I asked about supporting all media (and signaling) via port >> 443 is that some users are in firewalls that block all ports except 443 (or >> 80). >> >> >> On Mon, Aug 10, 2020 at 5:14 AM < >> freeswitch-users-request(a)lists.freeswitch.org> wrote: >> >>> ---------- Forwarded message ---------- >>> From: Sergey Safarov <s.safarov(a)gmail.com> >>> >>> You can offer ice for all calls from FreeSwitch without coturn >>> >>> <X-PRE-PROCESS cmd="set" data="add_ice_candidates=true"/> >>> >>> More details >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTrave… >>> >>> also, you can try the variable "media_webrtc=true" >>> >>> >>> >>> On Sun, Aug 9, 2020 at 8:42 PM David P <davidswalkabout(a)gmail.com> >>> wrote: >>> >>>> Can FS v1.10 be configured to handle SRTP of all calls via port 443? >>>> >>>> This would allow us to stop using coturn, which is often too slow to >>>> provide a TLS relay candidate before browsers finish gathering ice >>>> candidates. >>>> >>> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://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 >> https://freeswitch.com > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > ---------- Forwarded message ---------- > From: Prashant Lamba <prashant.lamba(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 12 Aug 2020 13:48:26 +0530 > Subject: Re: [Freeswitch-users] Change name shown on the ITSP or > registered users > You can do it in the SIP profile (under "user-agent-string"). But why > would you want to hide the fact that you're using FS :-))) > > Prashant > -- > To save our tigers, save their habitat. Think before you print this email. > <www.saveourtigers.com> > > On Tue, Aug 11, 2020 at 1:49 AM John Tuxies <atuxnull(a)gmail.com> wrote: > >> I would like to change the Freeswitch name shown when a users connects >> to or to the ITSP. So they will not know the name and version of FS. I >> would like to name it something like sip.lab.mydomain.org >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://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 >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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is sip_renegotiate_codec_on_reinvite still there?
by David Villasmil 09 Sep '20

09 Sep '20

Hello all, is https://freeswitch.org/confluence/display/FREESWITCH/sip_renegotiate_codec_… still alive? I can't find it in the source. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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