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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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output script as variable
by thomas peterseil 09 Sep '20

09 Sep '20

dear freeswitch-users, i have a very simple bash script and i would like to use the output of this script as a variable in the dialplan, but it doesn´t work. can somebody give me a hint why it doesn´t work. here is the script: hello.sh --------- #!/bin/bash echo "red.wav" ---- here is the relevant part of my dialplan: ---- <extension name="inbound_uri_test"> <condition field="destination_number" expression="^test$"> <action application="answer"/> <action application="set" data="col=${system /root/hello.sh}"/> <action application="log" data="INFO The value of the variable is '${col}'"/> <action application="playback" data="/root/audio/${col}"/> </condition> </extension> ------ i followed the documentation from this page: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+system thank you very much! best regards, thomas

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mod_portaudio on Raspberry Pi does not find connected audio devices
by Tazari, Mohammad Reza 07 Sep '20

07 Sep '20

Hi, here is a short summary of my configuration and issue: 1. Pi-4 with Raspbian buster and latest related FreeSwitch distribution, apt-upgraded and updated on last Friday 2. aplay is working both with the audio jack and with bluetooth (although not simultaneously: as soon as I have a bluetooth audio connection, the audio jack stops to work, and I cannot switch back to it with "amixer cset numid=3 1", but aplay can continue to use the bluetooth connection again after I issue "amixer cset numid=3 65536") 3. It seems that I do not have any permission issues as both of the users pi and freeswitch are in the audio group, which has rw access to the related devices under /dev/snd 4. I have already installed alsaoss, libasound2 and libasound2-dev 5. The output of "pa rescan" in fs_cli is the following: 2020-06-19 06:55:18.803478 [INFO] mod_portaudio.c:3186 Looking for new devices. 2020-06-19 06:55:18.803478 [INFO] mod_portaudio.c:2102 PortAudio version number = 1246720 PortAudio version text = 'PortAudio V19.6.0-devel, revision 396fe4b6699ae929d3a685b3ef8a7e97396139a4' 2020-06-19 06:55:18.943433 [DEBUG] mod_portaudio.c:1773 global indev [-1] 2020-06-19 06:55:18.943433 [ERR] mod_portaudio.c:1777 Cannot find an input device 2020-06-19 06:55:18.943433 [DEBUG] mod_portaudio.c:1786 global outdev [-1] 2020-06-19 06:55:18.943433 [ERR] mod_portaudio.c:1790 Cannot find an output device 2020-06-19 06:55:18.943433 [INFO] mod_portaudio.c:2125 Number of devices = 0 Any clue, what could be wrong on my side? Thanks! -- Saied

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cannot do bujild-deps on debian 10 for freeswitch.
by John Covici 03 Sep '20

03 Sep '20

Hi. I have Debian 10 now release 5 and cannot do apt-get build-deps for freeswitch. Also, I note the the keyring and archives still say stretch and there is none for buster. Here is what I get when trying to do build-deps: apt-get build-dep freeswitch Reading package lists... Done Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: builddeps:freeswitch : Depends: libpcre3-dev but it is not going to be installed Depends: libsofia-sip-ua-dev (>= 1.12.12) but it is not going to be installed Depends: libks but it is not going to be installed Depends: signalwire-client-c but it is not going to be installed Depends: libmagickcore-dev Depends: libglib2.0-dev but it is not going to be installed E: Unable to correct problems, you have held broken packages. I tried to do apt-get install libpcre3-dev and it still complained, seemed to want an older version of the packages than the one that was installed. How to fix? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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CNG Detection for T.38 <--> G.711 Transcoding
by Gabriel Kuri 31 Aug '20

31 Aug '20

Hi All: Trying to do T.38 <--> G.711 transcoding using "t38_gateway=self cng" in the dialplan. Here's the scenario... A-leg: Analog Telephone Adapter (T.38) FS in the middle B-leg: Out to Carrier/Provider (G.711u) The ATA initiates the call outbound to the carrier/provider. The far end FAX machine picks up and the call begins. I'm wanting FS to Re-INVITE T.38 to the ATA on the a-leg when it detects a CNG tone. That seems to work about 50% of the time (on average). The other 50% of the time, it doesn't seem to detect the FAX tone and the call stays as G.711u and eventually fails. Below is the dialplan for reference. I'm guessing the CNG tone detection on FS is hit and miss because of a timing issue, but I'm not really sure. Is it possible to detect the CED, V.21 Preamble, or /ANSAM of the receiving fax (far-end), in order to try and make the switchover more reliable? Or am I barking up the wrong tree here and/or possibly doing something wrong? For what it's worth, I've also tried changing the dialplan to have FS force the T.38 Re-INVITE "nocng" (ie execute_on_answer=t38_gateway self nocng), and FS Re-INVITEs T.38 immediately after the 200 OK. While the ATA seems to accept the T.38 Re-INVITE, it then stops sending RTP/UDPTL traffic entirely and the call eventually fails. It doesn't seem to like the Re-INVITE when it occurs that fast. I've also tried this on multiple ATAs with the same result. Thanks! <extension name="to_fax_dest"> <condition field="destination_number" expression="^\+\d{11}$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="t38_passthru=false"/> <action application="set" data="fax_verbose=true"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway self cng"/> <action application="bridge" data="{refuse_t38=true}sofia/sip/xxxxx(a)aa.bb.cc.dd"/> </condition> </extension>

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Re: [Freeswitch-users] FreeSWITCH suddenly have a child processes
by ziv＠ip-com.co.il 31 Aug '20

31 Aug '20

Hello � We have an occasional situation where the one FreeSWITCH process suddenly have a child process. We do not know what circumstances may have caused it. Happens every few days. The child process consumes memory like the normal process, which is a significant amount. It doesn’t consume CPU. In the case that we didn’t restart FreeSWITCH after the problem happened, later it happened again and we had 2 child processes. When killing the child process, FreeSWITCH seems to continue working normally, but this is not a thing we’d do on our production machines. Happens on one specific FreeSWITCH, our other machines do not show this behavior. The FreeSWITCH log for the time the child process starts shows initialization messages as when the normal process starts, but nothing that could tell us why the child starts. � After noticing the issue on FreeSWITCH 1.8.5, we upgraded to 1.10.5 via “apt” (didn’t build from source) but the problem is still there. � O/S: Debian 9.13 (we didn’t upgrade it when upgraded FreeSWITCH) FreeSWITCH: 1.10.5-release-17-25569c1631~64bit We use FusionPBX to operate the system. � “ps -ef” shows: UID PID PPID C STIME TTY TIME CMD freeswi+ 470 23710 0 11:25 ? 00:00:00 /usr/bin/freeswitch -ncwait -nonat freeswi+ 23710 1 62 Aug26 ? 1-11:31:26 /usr/bin/freeswitch -ncwait -nonat � “top” shows PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 23710 freeswi+ -2 19 3627008 1.543g 13972 S 110.7 20.7 2136:50 freeswitch 470 freeswi+ 39 19 3615296 1.507g 884 S 0.0 20.2 0:00.02 freeswitch � An advice will be appreciated, Ziv

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