From david.villasmil.work at gmail.com Tue Sep 1 12:43:14 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 1 Sep 2020 13:43:14 +0100 Subject: [Freeswitch-users] is sip_renegotiate_codec_on_reinvite still there? Message-ID: Hello all, is https://freeswitch.org/confluence/display/FREESWITCH/sip_renegotiate_codec_on_reinvite still alive? I can't find it in the source. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Wed Sep 2 18:54:34 2020 From: covici at ccs.covici.com (John Covici) Date: Wed, 02 Sep 2020 14:54:34 -0400 Subject: [Freeswitch-users] cannot do bujild-deps on debian 10 for freeswitch. Message-ID: 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 at ccs.covici.com From brian at freeswitch.com Thu Sep 3 02:44:45 2020 From: brian at freeswitch.com (Brian West) Date: Wed, 2 Sep 2020 21:44:45 -0500 Subject: [Freeswitch-users] cannot do bujild-deps on debian 10 for freeswitch. In-Reply-To: References: Message-ID: Those all exist in our repo. Make sure to add ours to your sources.lst On Wed, Sep 2, 2020 at 2:38 PM John Covici wrote: > 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 at ccs.covici.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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Thu Sep 3 09:11:25 2020 From: covici at ccs.covici.com (John Covici) Date: Thu, 03 Sep 2020 05:11:25 -0400 Subject: [Freeswitch-users] cannot do bujild-deps on debian 10 for freeswitch. In-Reply-To: References: Message-ID: I changed my freeswitch packages to debian-unstable which fixed some of the unmet dependencies, but still getting the following: When I itry to do apt-get build-dep freeswitch I get: 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: libmagickcore-dev Depends: libglib2.0-dev but it is not going to be installed E: Unable to correct problems, you have held broken packages. In an older version of Debian buster (release 3 on a different system) I have done build-dep successfully, but I cannot do this in the current Debian buster 10.5, what can I do? On Wed, 02 Sep 2020 22:44:45 -0400, Brian West wrote: > > [1 ] > [1.1 ] > Those all exist in our repo. Make sure to add ours to your sources.lst > > > On Wed, Sep 2, 2020 at 2:38 PM John Covici wrote: > > > 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 at ccs.covici.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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > [1.2 ] > [2 ] > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From davidswalkabout at gmail.com Fri Sep 4 02:07:46 2020 From: davidswalkabout at gmail.com (David P) Date: Fri, 4 Sep 2020 14:07:46 +1200 Subject: [Freeswitch-users] Handle media of all calls via port 443? In-Reply-To: References: Message-ID: Hi Brian W, Back on Aug 10, Sergey Safarov suggested we might get FS's answer sdp to include more candidates via variable "media_webrtc=true". (We'd like TCP via port 443.) The only documentation I can find is an old post by you: http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html It didn't work for me when I set "media_webrtc" as a regular dialplan var, but your post mentions the {} of a bridge. We use a conference, so should we set it this way? On Thu, Aug 13, 2020 at 6:21 AM David P wrote: > 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 at lists.freeswitch.org> wrote: > >> ---------- Forwarded message ---------- >> From: Giovanni Maruzzelli >> >> In this case, you *must* restart freeswitch, reloadxml is not enough >> -giovanni >> >> On Tue, Aug 11, 2020 at 9:01 PM David P >> 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 >>> in >>> /etc/freeswitch/vars.xml >>> and >>> 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 at lists.freeswitch.org> wrote: >>> >>>> ---------- Forwarded message ---------- >>>> From: Sergey Safarov >>>> >>>> You can offer ice for all calls from FreeSwitch without coturn >>>> >>>> >>>> >>>> More details >>>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal#NATTraversal-FreeSWITCHBehindNAT >>>> >>>> also, you can try the variable "media_webrtc=true" >>>> >>>> >>>> >>>> On Sun, Aug 9, 2020 at 8:42 PM David P >>>> 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 at 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 at 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 >> To: FreeSWITCH Users Help >> 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. >> >> On Tue, Aug 11, 2020 at 1:49 AM John Tuxies 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From saied.tazari at gmail.com Fri Sep 4 21:42:29 2020 From: saied.tazari at gmail.com (Tazari, Mohammad Reza) Date: Fri, 4 Sep 2020 23:42:29 +0200 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: <175ee000-d1d7-1eac-7032-da1c1cd6f547@gmail.com> References: <477e6aac-f48f-3dcd-59d0-a94a748758ca@gmail.com> <175ee000-d1d7-1eac-7032-da1c1cd6f547@gmail.com> Message-ID: <22413535-5003-15db-782f-e385336b9c73@gmail.com> Hi, I have to correct myself with regard to my two posts on Aug 21: 1. I was testing with my headset and had successful "pa looptest" indicating both indev & outdev were working 2. Then, when I tested that in the context of my application for auto-answering incoming calls (I called with my mobile phone while putting it in front of a tv so that I can test if I would hear anything on my headset), I didn't notice that in that case the mic was not working because I had the headset on my ears and was hearing the tv (because of successful looptest, I was just assuming that the mic would also be working) 3. But when I finally got a Plantronics Calisto 620-M Bluetooth Speakerphone, even the looptest failed. At this moment, I tested more carefully my headset again, and there I found out that the mic was not working. 4. Because it seemed that something device-specific should be the reason for failure, I also tested with my bluetooth earphones; there the effect was equivalent to Calisto speakerphone. 5. I finally got a USB speakerphone and tried that one, but in that case I was again back to earlier situation where "pa rescan" failed to find any devices... With all four devices, always arecord and aplay were working perfectly, and the above issues appeared only with mod_portaudio. I don't have any idea what the issue could be with the usb, but with the three bluetooth devices, I still believe that something device-specific should be causing the problems here, e.g. the sample rate. But, currently I am lacking time to further work on this... KR, -- Saied Tazari, Mohammad Reza wrote on 21-Aug-20 18:44: > Tazari, Saied wrote on 21-Aug-20 03:37: >> >> 6. Created a new file by _*sudo vi /etc/asound.conf*_ and added the >> following as its sole content (if you know the MAC address of >> your Bluetooth device, substitute it already now; otherwise you >> can change it also later in step 9): >> > > This was a very important reason why "pa rescan " had failed for me > previously: I had defined the devices in the .asoundrc of the user pi, > but freeswitch was running with the user "freeswitch" who has no login > shell and hence cannot have any .asoundrc --> device configuration had > to go to a user-independent level. > > The suggested content for this file tried to get a better output > quality by making use of the "a2dp" profile whereas for the mic you > have to use the "sco" profile. However, in this combination pa is > eventually working in a pretty unstable way. Therefore, we can > simplify the content the following way: > > pcm.btAudioIO { >         type plug >         slave.pcm { >                 type bluealsa >                 service "org.bluealsa" >                 device "00:11:09:94:14:3D" >                 profile "sco" >                 delay 10000 >         } >         hint { >                 show on >                 description "Bluetooth Audio Input/Output Dvice" >         } > } > > ctl.btAudioIO { >         type bluealsa >         service "org.bluealsa" >         battery yes > } > >> 10. Test your device with the following two commands (five seconds >> recording): >> *arecord -D btAudioIO -d 5 test.wav >> aplay ***-D btAudioIO test.wav >> ** >> > > You may also perform a loop test: > > *arecord -D btAudioIO | aplay -D btAudioIO* > >> After a reboot, after which also FreeSwitch and my application start, >> everything was working for me just fine :-) >> > > I forgot to say that you can test the working of the setup in fs_cli > the following way: > > 1. "*pa devlist*" should generate an output like > "*0;btAudioIO(ALSA);128;128;r,i,o*" > 2. Like the test at the level of shell, the actual test can be the > loop test: "*pa looptest*" > > > KR, > > > -- Saied > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Mon Sep 7 06:24:04 2020 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Mon, 7 Sep 2020 06:24:04 +0000 Subject: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP In-Reply-To: <450B9BDB-ED7D-44BE-A2BE-AF783DE6D369@freeswitch.org> References: , <450B9BDB-ED7D-44BE-A2BE-AF783DE6D369@freeswitch.org> Message-ID: Hi All, I raised the issue with IETF dispatch and had the following response from Paul Kyzivat, based on this response I believe the correct default behaviour according to RFC (as it is now) is that we should be offering an SDP with 'a=sendrecv' in response to a RE-INVITE with-out SDP (3PCC) rather than carrying over 'a=sendonly' from the existing session. On 9/5/20 4:10 PM, Shaun Stokes wrote: > Hi Paul, > > Thanks for your response. > > RFC 6337 section 5.1 refers us back to RFC 3261 in case of a RE-INVITE > and "without regard for what the other party in the call may have > indicated previously" would suggest we should be using 'a=sendrecv' in > our offer. As one of the authors of 6337 I will agree that sendrecv is probably what the UAS should be offering given the circumstances. But it ultimately comes down to what it "wants" to be doing at that time. The folly comes when it offers something less than what *it* wants because it imagines (based on prior o/a) that the answerer wants less than it does. This can get you into "stuck on hold" scenarios or other trouble. Thanks, Paul > I previously tried to touch base with Henning and was directed to the > dispatch mail list. > > I'll post the question to the sip-implementors mail list. > > Thanks, > Shaun > ------------------------------------------------------------------------ > *From:* Paul Kyzivat > *Sent:* 05 September 2020 17:16 > *To:* Shaun Stokes > *Cc:* dispatch at ietf.org > *Subject:* Re: [dispatch] RFC 3261 section 14.2 - "brand new call" does > not specify whether the SDP should modify media attributes of an > existing session containing a=sendonly or a=recvonly > Shaun, > > Take a look at RFC6337 (especially section 5.1) and see if it helps. > That RFC was written to respond to many questions about O/A that came up > over time. It is not normative, but rather simply clarifies things that > are implicit upon analyzing an assortment of normative RFCs. > > BTW, dispatch isn't really the right place for a question like this. A > better place is . > > Thanks, > Paul I have also raised this with SIP Implementors as this contradicts a previous discussion on this issue involving the authors of RFC 3261 (http://marc.info/?t=98738614300001&r=1&w=2). Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Mike Jerris Sent: 31 August 2020 17:46 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP Yes sip_unhold_nosdp does exactly what you are asking. On Aug 28, 2020, at 6:05 AM, Shaun Stokes > wrote: 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: From tom at tomlynn.com Mon Sep 7 16:06:07 2020 From: tom at tomlynn.com (Tom Lynn) Date: Mon, 7 Sep 2020 09:06:07 -0700 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: References: <477e6aac-f48f-3dcd-59d0-a94a748758ca@gmail.com> Message-ID: 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? ---------- Forwarded message --------- From: Tazari, Mohammad Reza Date: Wed, Aug 26, 2020 at 11:01 AM Subject: Re: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? To: Tom, other interested people, I finally got it working on a raspberry pi-4 with buster lite, mostly thanks to the following articles: https://sigmdel.ca/michel/ha/rpi/bluetooth_01_en.html https://sigmdel.ca/michel/ha/rpi/bluetooth_02_en.html https://sigmdel.ca/michel/ha/rpi/bluetooth_n_buster_01_en.html The exact reproducible steps have been: 1. Started on the basis of + my FreeSwitch installation based on instructions under https://freeswitch.org/confluence/display/FREESWITCH/Raspberry+Pi 2. *sudo apt-get update && sudo apt-get install -y bluealsa * 3. *sudo vi /lib/systemd/system/bluealsa.service* --> substituted ExecStart=/usr/bin/bluealsa with --> Environment=LIBASOUND_THREAD_SAFE=0 ExecStart=/usr/bin/bluealsa -p hfp-hf -p hsp-hs -p hfp-ag -p hsp-ag -p a2dp-source -p a2dp-sink 4. *sudo vi /lib/systemd/system/bluetooth.service* --> Added --noplugin=sap to the end of the following line ExecStart=/usr/lib/bluetooth/bluetoothd 5. *sudo vi /lib/systemd/system/bthelper at .service* --> Added ExecStartPre=/bin/sleep 2 in a new line after Type=simple 6. Created a new file by *sudo vi /etc/asound.conf* and added the following as its sole content (if you know the MAC address of your Bluetooth device, substitute it already now; otherwise you can change it also later in step 9): pcm.btAudioIO { type asym playback.pcm { type plug slave.pcm { type bluealsa service "org.bluealsa" device "00:11:09:94:14:3D" profile "a2dp" delay 10000 } } capture.pcm { type plug slave.pcm { type bluealsa service "org.bluealsa" device "00:11:09:94:14:3D" profile "sco" delay 10000 } } hint { show on description "Bluetooth Audio Input/Output Dvice" } } 7. *sudo adduser freeswitch audio sudo adduser * *freeswitch bluetooth sudo reboot * 8. After reboot, run *sudo bluetoothctl* and then find, pair, connect and trust your Bluetooth device with the following control commands of this program: *agent on scan on* <-- find the MAC address of your device in the scan outputs * pair 50:11:2D:06:ED:2F * *scan off * *connect 50:11:2D:06:ED:2F trust 50:11:2D:06:ED:2F exit *--> from now on, your device should automatically reconnect to the pi each time after the pi is rebooted; if, however, this is not the case, you will have to redo this step after each reboot. 9. In this example, the device that I have paired and trusted has a different MAC address as the one in /etc/asound.conf; therefore we must edit the file and set the right MAC address. If you already had the right MAC address, you may go to the next step without reboot; but, if you had to change the file /etc/asound.conf, then you will have to reboot before going to the next step. 10. Test your device with the following two commands (five seconds recording): *arecord -D btAudioIO -d 5 test.wav aplay * *-D btAudioIO test.wav * 11. Assuming that the above test is successful, you now - edit /autoload_configs/modules.conf.xml and un-comment the loading of mod_portaudio - edit /autoload_configs/portaudio.conf.xml and substitute the whole content with the following new content: c 12. In my case, I didn't need to do anything for dial plans because in my system both the in- and outbound calls are first handled by my python scripts; in particular, the BT handsfree device is never supposed to make calls but is called for making announcements or invited to a running call as the third leg. So far, I was using an IP telephone with auto-answering capability as extension 000 of my FreeSwitch for this purpose. To substitute that phone with the BT handsfree device, I just needed to update my related python script with the following command: *sudo sed -i 's#user/000@[0-9.]+#portaudio/auto_answer#g' dial.py* After a reboot, after which also FreeSwitch and my application start, everything was working for me just fine :-) Cheers, -- Saied ForwardedMessage.eml Subject: Re: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? From: Tom Lynn Date: 27-Jul-20, 07:56 To: FreeSWITCH Users Help I'm in the same boat with portaudio when it's started with systemd. When I disable it as a service and start it from the command line, pa devlist does return a list of devices. Still workingon getting a call from portaudio to a station and vice versa. Calling a station from portaudio device goes to voicemail and in reverse direction I never get a ring indication, just music on hold. On Tue, Jun 30, 2020 at 8:21 PM saiedt wrote: Lesley Pervis wrote > Sorry for the spam, but I thought I'd mention that I can aplay no problem > from the command line as the 'freeswitch' user, and yet when FS is running > as that user, no luck. Exactly seven years after your last post here, I'm there to ask if you finally succeeded to resolve the issue? My case: 1. Pi-4 with Raspbian buster und latest related FreeSwitch distribution, apt-upgraded and updated just yesterday 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 issue as both pi and freeswitch are in the audio group and related devices under /dev/snd are in that group 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! _________________________________________________________________________ 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ipmekgmohbekpikl.png Type: image/png Size: 9299 bytes Desc: not available URL: From saied.tazari at gmail.com Tue Sep 8 09:32:29 2020 From: saied.tazari at gmail.com (Tazari, Mohammad Reza) Date: Tue, 8 Sep 2020 11:32:29 +0200 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 171, Issue 7 In-Reply-To: References: Message-ID: <0cd9f1f6-1c12-c063-0c77-544f78433f14@gmail.com> 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+Tasks 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 at 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 > Date: > 07-Sep-20, 18:06 > > To: > FreeSWITCH Users Help > > > 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Tue Sep 8 16:46:06 2020 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 8 Sep 2020 21:46:06 +0500 Subject: [Freeswitch-users] [In Progress/Ringing Calls][FreeSWITCH Event] Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Tue Sep 8 21:33:20 2020 From: davidswalkabout at gmail.com (David P) Date: Wed, 9 Sep 2020 09:33:20 +1200 Subject: [Freeswitch-users] How to avoid clipping when transcoding to PCMA/U ? Message-ID: 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Tue Sep 8 22:51:31 2020 From: davidswalkabout at gmail.com (David P) Date: Wed, 9 Sep 2020 10:51:31 +1200 Subject: [Freeswitch-users] How to avoid clipping when transcoding to PCMA/U ? In-Reply-To: References: Message-ID: Our conference.conf.xml had: and no auto-gain-level. I changed it: And there's now clipping only during the loudest parts. But I think this gain setting is already extreme and I'd like to know if there's a less brute-force way to eliminate clipping. On Wed, Sep 9, 2020 at 9:33 AM David P wrote: > 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? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 9 00:09:42 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 9 Sep 2020 01:09:42 +0100 Subject: [Freeswitch-users] issue with mod_python Message-ID: 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": 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 at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 9 00:15:25 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 9 Sep 2020 01:15:25 +0100 Subject: [Freeswitch-users] issue with mod_python In-Reply-To: References: Message-ID: Hello all, And that's the script directory: # fs_cli -x 'global_getvar'| grep _dir base_dir=/usr recordings_dir=/var/lib/freeswitch/recordings sounds_dir=/usr/share/freeswitch/sounds conf_dir=/etc/freeswitch log_dir=/var/log/freeswitch run_dir=/var/run/freeswitch db_dir=/var/lib/freeswitch/db mod_dir=/usr/lib/freeswitch/mod htdocs_dir=/usr/share/freeswitch/htdocs *script_dir=/usr/share/freeswitch/scripts* temp_dir=/tmp grammar_dir=/usr/share/freeswitch/grammar fonts_dir=/usr/share/freeswitch/fonts images_dir=/var/lib/freeswitch/images certs_dir=/etc/freeswitch/tls storage_dir=/var/lib/freeswitch/storage cache_dir=/var/cache/freeswitch data_dir=/usr/share/freeswitch localstate_dir=/var/lib/freeswitch Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Sep 9, 2020 at 1:09 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > 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": > 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 at gmail.com > phone: +34669448337 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Sep 9 00:47:54 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 9 Sep 2020 01:47:54 +0100 Subject: [Freeswitch-users] issue with mod_python In-Reply-To: References: Message-ID: Hello all, Nevermind everyone, i was missing some modules. Thanks! Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Sep 9, 2020 at 1:15 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello all, > > And that's the script directory: > > # fs_cli -x 'global_getvar'| grep _dir > > base_dir=/usr > recordings_dir=/var/lib/freeswitch/recordings > sounds_dir=/usr/share/freeswitch/sounds > conf_dir=/etc/freeswitch > log_dir=/var/log/freeswitch > run_dir=/var/run/freeswitch > db_dir=/var/lib/freeswitch/db > mod_dir=/usr/lib/freeswitch/mod > htdocs_dir=/usr/share/freeswitch/htdocs > *script_dir=/usr/share/freeswitch/scripts* > temp_dir=/tmp > grammar_dir=/usr/share/freeswitch/grammar > fonts_dir=/usr/share/freeswitch/fonts > images_dir=/var/lib/freeswitch/images > certs_dir=/etc/freeswitch/tls > storage_dir=/var/lib/freeswitch/storage > cache_dir=/var/cache/freeswitch > data_dir=/usr/share/freeswitch > localstate_dir=/var/lib/freeswitch > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Wed, Sep 9, 2020 at 1:09 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> 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": >> 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 at gmail.com >> phone: +34669448337 >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun at sysconfig.cloud Wed Sep 9 06:55:16 2020 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Wed, 9 Sep 2020 06:55:16 +0000 Subject: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP In-Reply-To: References: , <450B9BDB-ED7D-44BE-A2BE-AF783DE6D369@freeswitch.org>, Message-ID: We've had a response from Roman Shpount on SIP Implementors who "got the language about handing re-INVITE without SDP into RFC 3261 as a brand new call", conversation below. I'll raise this as an issue on GitHub when I have some time to build and test on master. I've also noticed FreeSWITCH doesn't respond to "a=sendonly" with "a=recvonly" possibly because the ACK SDP from 3rd party was immediately followed with a RE-INVITE with-out SDP so FreeSWITCH carries over the ACK SDP with no change. Hi Roman, Thank you for your response. We are using FreeSWITCH as a SIP and RTP media server to connect the caller (leg a) to the callee (leg b), the caller is expecting the media state sendrecv but this is influenced by the 3rd party. The call flow is as follows. -> = generated by FreeSWITCH (caller) <- = generated by 3rd party (callee) -> 0.000000s INVITE with SDP 'codec list' <- 0.012557s 100 Trying <- 0.083254s 200 OK with SDP 'codec list' -> 0.085348s ACK <- 0.071315s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.086461s 100 Trying -> 0.087391s 200 OK with SDP 'codec list' <- 0.154249s ACK with SDP 'codec list and a=sendonly' <- 0.155111s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.166856s 100 Trying -> 0.167631s 200 OK with SDP 'codec list and a=sendonly' <- 0.202331s ACK with SDP 'codec list and a=recvonly' <- 0.337532s INVITE with-out SDP 'RE-INVITE for existing session' -> 0.346448s 100 Trying -> 0.347170s 200 OK with SDP 'codec list and a=sendonly' <- 0.390116s ACK with SDP 'codec list and a=recvonly' FreeSWITCH currently interprets a RE-INVITE with-out SDP for an existing session as 'no change' for the hold state so it's carrying 'a=sendonly' over from the existing session as it was in the ACK SDP generated by the 3rd party. Based on your explanation I believe this is wrong and we should be responding with-out 'a=sendonly' (default behaviour) or with 'a=sendrecv'. Thanks, Shaun ________________________________ From: Roman Shpount Sent: 07 September 2020 08:44 To: Shaun Stokes Cc: sip-implementors at lists.cs.columbia.edu Subject: Re: [Sip-implementors] RFC 3261 section 14.2 - "brand new call" does not specify whether the SDP should modify media attributes of an existing session containing a=sendonly or a=recvonly Shaun, I am the person who actually got the language about handing re-INVITE without SDP into RFC 3261 as "a brand new call". The initial intent was to enable a third party call control to initiate a new call by sending a re-INVITE without SDP to an existing call and then place another call to a new party. If I understand correctly, FreeSwitch is sending a response with "a=recvonly" to a re-INVITE with no SDP? If this is the case, since they are a media server, in this particular situation they are probably wrong, but generally the answer is "it depends". Because of this, you are not going to find an RFC that specifies the one and only correct procedure. The general idea is that sendonly/recvonly in every SDP exchange should reflect the preferences for the user agents, not what was previously negotiated. Imagine that one UA is putting another UA on hold. In this case this phone sends a re-INVITE with a=inactive (or a=sendonly which only makes sense if the UA plans to play the music on hold). The second UA will respond with a=inactive or a=recvonly. If the second UA later sends a re-INVITE without SDP, the first UA will still respond with SDP with a=inactive (or a=sendonly), since it is still on hold. If the UA which is currently on hold sends a re-INVITE with no SDP, then the other UA should respond with a=sendrecv (since it is not on hold), but the first UA should respond with a=inactive (or a=sendonly) in SDP in ACK, since it is still on hold. In other words, re-INVITE does not change the local UA hold status, only a user action does this. Based on the local hold status and the remote direction attribute the UA should respond with an appropriate direction attribute in the answer. If you are using FreeSwitch as a media server, then the local call status is likely not on hold and it should be able to send/recv media, which should be indicated in the response to a re-INVITE with no body. In general case, the local call status is something that depends on the application running on FreeSwitch, which you do not specify. This is why the general answer "it depends". I hope it helps, _____________ Roman Shpount ________________________________ From: FreeSWITCH-users on behalf of Shaun Stokes Sent: 07 September 2020 08:24 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP Hi All, I raised the issue with IETF dispatch and had the following response from Paul Kyzivat, based on this response I believe the correct default behaviour according to RFC (as it is now) is that we should be offering an SDP with 'a=sendrecv' in response to a RE-INVITE with-out SDP (3PCC) rather than carrying over 'a=sendonly' from the existing session. On 9/5/20 4:10 PM, Shaun Stokes wrote: > Hi Paul, > > Thanks for your response. > > RFC 6337 section 5.1 refers us back to RFC 3261 in case of a RE-INVITE > and "without regard for what the other party in the call may have > indicated previously" would suggest we should be using 'a=sendrecv' in > our offer. As one of the authors of 6337 I will agree that sendrecv is probably what the UAS should be offering given the circumstances. But it ultimately comes down to what it "wants" to be doing at that time. The folly comes when it offers something less than what *it* wants because it imagines (based on prior o/a) that the answerer wants less than it does. This can get you into "stuck on hold" scenarios or other trouble. Thanks, Paul > I previously tried to touch base with Henning and was directed to the > dispatch mail list. > > I'll post the question to the sip-implementors mail list. > > Thanks, > Shaun > ------------------------------------------------------------------------ > *From:* Paul Kyzivat > *Sent:* 05 September 2020 17:16 > *To:* Shaun Stokes > *Cc:* dispatch at ietf.org > *Subject:* Re: [dispatch] RFC 3261 section 14.2 - "brand new call" does > not specify whether the SDP should modify media attributes of an > existing session containing a=sendonly or a=recvonly > Shaun, > > Take a look at RFC6337 (especially section 5.1) and see if it helps. > That RFC was written to respond to many questions about O/A that came up > over time. It is not normative, but rather simply clarifies things that > are implicit upon analyzing an assortment of normative RFCs. > > BTW, dispatch isn't really the right place for a question like this. A > better place is . > > Thanks, > Paul I have also raised this with SIP Implementors as this contradicts a previous discussion on this issue involving the authors of RFC 3261 (http://marc.info/?t=98738614300001&r=1&w=2). Thanks, Shaun ________________________________ From: FreeSWITCH-users on behalf of Mike Jerris Sent: 31 August 2020 17:46 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP Yes sip_unhold_nosdp does exactly what you are asking. On Aug 28, 2020, at 6:05 AM, Shaun Stokes > wrote: 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? - T : E : W : www.sysconfig.cloud SYSCONFIG is a trading name of ITEC Support LTD which is a limited company registered in England and Wales. Company Registered Number 06908001. Registered office: Suite 2, Prospect House, Bath Road Trading Estate, Stroud, GL5 3QF. 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URL: From thomas.peterseil at mine-project.eu Wed Sep 9 18:38:19 2020 From: thomas.peterseil at mine-project.eu (thomas peterseil) Date: Wed, 9 Sep 2020 20:38:19 +0200 Subject: [Freeswitch-users] output script as variable Message-ID: 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: ---- ------ i followed the documentation from this page: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+system thank you very much! best regards, thomas From olle at zaark.com Wed Sep 2 12:20:59 2020 From: olle at zaark.com (Olle Frimanson) Date: Wed, 2 Sep 2020 14:20:59 +0200 Subject: [Freeswitch-users] Early media in the "wrong" direction Message-ID: <00de01d68123$84cd16c0$8e674440$@zaark.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Sep 9 20:20:12 2020 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 9 Sep 2020 16:20:12 -0400 Subject: [Freeswitch-users] is sip_renegotiate_codec_on_reinvite still there? In-Reply-To: References: Message-ID: commit 2e3227b50f897ab7471b09979ba25b6ca0ff5887 Author: Anthony Minessale Date: Fri Aug 12 14:10:23 2016 -0500 FS-9422 #resolve [Freeswitch Exit/Crash on SDP Negotiation] #comment renegotiate-codec-on-hold renegotiate-codec-on-reinvite are both removed in this commit > On Sep 1, 2020, at 8:43 AM, David Villasmil wrote: > > Hello all, > > is https://freeswitch.org/confluence/display/FREESWITCH/sip_renegotiate_codec_on_reinvite still alive? I can't find it in the source. > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Sep 9 20:22:18 2020 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 9 Sep 2020 16:22:18 -0400 Subject: [Freeswitch-users] Handle media of all calls via port 443? In-Reply-To: References: Message-ID: FS won’t but a client that supports turn could use that via the turn server. > On Sep 3, 2020, at 10:07 PM, David P wrote: > > Hi Brian W, > > Back on Aug 10, Sergey Safarov suggested we might get FS's answer sdp to include more candidates via variable "media_webrtc=true". (We'd like TCP via port 443.) The only documentation I can find is an old post by you: > http://lists.freeswitch.org/pipermail/freeswitch-users/2015-September/115727.html > > It didn't work for me when I set "media_webrtc" as a regular dialplan var, but your post mentions the {} of a bridge. We use a conference, so should we set it this way? > > data="['conference_member_flags=endconf,jitterbuffer_msec=5p:100p,media_webrtc=true']sofia/gateway/{hidden}"/> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Sep 9 20:23:57 2020 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 9 Sep 2020 16:23:57 -0400 Subject: [Freeswitch-users] [In Progress/Ringing Calls][FreeSWITCH Event] In-Reply-To: References: Message-ID: <58EB7EC4-BBFA-4F58-A749-99B2C3474F28@freeswitch.org> You may not always get progress events depending on the signaling. It can go from new to answered without progress. > On Sep 8, 2020, at 12:46 PM, Bilal Abbasi wrote: > > 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. From mike at freeswitch.org Wed Sep 9 20:28:13 2020 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 9 Sep 2020 16:28:13 -0400 Subject: [Freeswitch-users] Late Codec Negotiation between FreeSWITCH and Broadsoft equipment when responding to RE-INVITE without SDP In-Reply-To: References: <450B9BDB-ED7D-44BE-A2BE-AF783DE6D369@freeswitch.org> Message-ID: <0A6E224E-ECF9-4EE2-A0B1-42D23A907ADA@freeswitch.org> At this point I think it makes sense to change default behavior here to match what it is with that variable. Feel free to make a pull request to make that change. > On Sep 9, 2020, at 2:55 AM, Shaun Stokes wrote: > > We've had a response from Roman Shpount on SIP Implementors who "got the language about handing re-INVITE without SDP into RFC 3261 as a brand new call", conversation below. I'll raise this as an issue on GitHub when I have some time to build and test on master. I've also noticed FreeSWITCH doesn't respond to "a=sendonly" with "a=recvonly" possibly because the ACK SDP from 3rd party was immediately followed with a RE-INVITE with-out SDP so FreeSWITCH carries over the ACK SDP with no change. > > Hi Roman, > > Thank you for your response. > > We are using FreeSWITCH as a SIP and RTP media server to connect the caller (leg a) to the callee (leg b), the caller is expecting the media state sendrecv but this is influenced by the 3rd party. > > The call flow is as follows. > > -> = generated by FreeSWITCH (caller) > <- = generated by 3rd party (callee) > > -> 0.000000s INVITE with SDP 'codec list' > <- 0.012557s 100 Trying > <- 0.083254s 200 OK with SDP 'codec list' > -> 0.085348s ACK > > <- 0.071315s INVITE with-out SDP 'RE-INVITE for existing session' > -> 0.086461s 100 Trying > -> 0.087391s 200 OK with SDP 'codec list' > <- 0.154249s ACK with SDP 'codec list and a=sendonly' > > <- 0.155111s INVITE with-out SDP 'RE-INVITE for existing session' > -> 0.166856s 100 Trying > -> 0.167631s 200 OK with SDP 'codec list and a=sendonly' > <- 0.202331s ACK with SDP 'codec list and a=recvonly' > > <- 0.337532s INVITE with-out SDP 'RE-INVITE for existing session' > -> 0.346448s 100 Trying > -> 0.347170s 200 OK with SDP 'codec list and a=sendonly' > <- 0.390116s ACK with SDP 'codec list and a=recvonly' > > FreeSWITCH currently interprets a RE-INVITE with-out SDP for an existing session as 'no change' for the hold state so it's carrying 'a=sendonly' over from the existing session as it was in the ACK SDP generated by the 3rd party. Based on your explanation I believe this is wrong and we should be responding with-out 'a=sendonly' (default behaviour) or with 'a=sendrecv'. > > Thanks, > Shaun > From: Roman Shpount > > Sent: 07 September 2020 08:44 > To: Shaun Stokes > > Cc: sip-implementors at lists.cs.columbia.edu > > Subject: Re: [Sip-implementors] RFC 3261 section 14.2 - "brand new call" does not specify whether the SDP should modify media attributes of an existing session containing a=sendonly or a=recvonly > > Shaun, > > I am the person who actually got the language about handing re-INVITE without SDP into RFC 3261 as "a brand new call". The initial intent was to enable a third party call control to initiate a new call by sending a re-INVITE without SDP to an existing call and then place another call to a new party. > > If I understand correctly, FreeSwitch is sending a response with "a=recvonly" to a re-INVITE with no SDP? If this is the case, since they are a media server, in this particular situation they are probably wrong, but generally the answer is "it depends". Because of this, you are not going to find an RFC that specifies the one and only correct procedure. The general idea is that sendonly/recvonly in every SDP exchange should reflect the preferences for the user agents, not what was previously negotiated. > > Imagine that one UA is putting another UA on hold. In this case this phone sends a re-INVITE with a=inactive (or a=sendonly which only makes sense if the UA plans to play the music on hold). The second UA will respond with a=inactive or a=recvonly. If the second UA later sends a re-INVITE without SDP, the first UA will still respond with SDP with a=inactive (or a=sendonly), since it is still on hold. If the UA which is currently on hold sends a re-INVITE with no SDP, then the other UA should respond with a=sendrecv (since it is not on hold), but the first UA should respond with a=inactive (or a=sendonly) in SDP in ACK, since it is still on hold. > > In other words, re-INVITE does not change the local UA hold status, only a user action does this. Based on the local hold status and the remote direction attribute the UA should respond with an appropriate direction attribute in the answer. If you are using FreeSwitch as a media server, then the local call status is likely not on hold and it should be able to send/recv media, which should be indicated in the response to a re-INVITE with no body. In general case, the local call status is something that depends on the application running on FreeSwitch, which you do not specify. This is why the general answer "it depends". > > I hope it helps, > _____________ > Roman Shpount > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Sep 9 20:29:10 2020 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 9 Sep 2020 16:29:10 -0400 Subject: [Freeswitch-users] Early media in the "wrong" direction In-Reply-To: <00de01d68123$84cd16c0$8e674440$@zaark.com> References: <00de01d68123$84cd16c0$8e674440$@zaark.com> Message-ID: <86FACA61-780F-4BA9-9115-E5728E70558B@freeswitch.org> Early media door phone like behavior like this is not currently supported. > On Sep 2, 2020, at 8:20 AM, Olle Frimanson wrote: > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From tom at tomlynn.com Thu Sep 10 02:32:03 2020 From: tom at tomlynn.com (Tom Lynn) Date: Wed, 9 Sep 2020 19:32:03 -0700 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 171, Issue 7 In-Reply-To: <0cd9f1f6-1c12-c063-0c77-544f78433f14@gmail.com> References: <0cd9f1f6-1c12-c063-0c77-544f78433f14@gmail.com> Message-ID: My unit file switches to user/group freeswitch. As such it does not return a result for "pa devlist." That only happens if I start freeswitch from the command line using sudo as root. Freeswitch user doesn't have a shell and even if it did, I suspect switching to that user would yield the same result. Swapping pi/pi in for the user and group in the unit file also results in the same no response. Only user=root context is successful in yielding a device list, as you say. ON this raspberry pi, I have only the 3.5mm jack and the bluetooth devices. Following your instructions, the bluetooth works and is more convenient for testing than wired headphones in the jack. I'm still hopeful that this can be figured out. Tom On Tue, Sep 8, 2020 at 2:33 AM Tazari, Mohammad Reza wrote: > 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+Tasks > 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 at 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 > > Date: > 07-Sep-20, 18:06 > > To: > FreeSWITCH Users Help > > 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? > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Sep 10 10:50:07 2020 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 10 Sep 2020 13:50:07 +0300 Subject: [Freeswitch-users] How to avoid clipping when transcoding to PCMA/U ? In-Reply-To: References: Message-ID: Does it clip when bridging too ? So not with mod_conference. You could try to add something like this in mod_opus.c, before decoding, and see if it helps: opus_decoder_ctl(context->decoder_object, OPUS_SET_GAIN(-10)); On Wed, Sep 9, 2020 at 1:52 AM David P wrote: > Our conference.conf.xml had: > > > > and no auto-gain-level. I changed it: > > > > > And there's now clipping only during the loudest parts. But I think this > gain setting is already extreme and I'd like to know if there's a less > brute-force way to eliminate clipping. > > On Wed, Sep 9, 2020 at 9:33 AM David P wrote: > >> 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? >> > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Thu Sep 10 17:07:32 2020 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 10 Sep 2020 22:07:32 +0500 Subject: [Freeswitch-users] [In Progress/Ringing Calls][FreeSWITCH Event] In-Reply-To: <58EB7EC4-BBFA-4F58-A749-99B2C3474F28@freeswitch.org> References: <58EB7EC4-BBFA-4F58-A749-99B2C3474F28@freeswitch.org> Message-ID: Thanks Mike, I was thinking same, but my wish list is a socket where we can see calls states(init, progress, media, answer, hangup). We do have events with all those status, but cant figure out to handle it efficiently when we have thousands of calls going out. Regards Abbasi On Thu, 10 Sep 2020 at 3:00 AM, Mike Jerris wrote: > You may not always get progress events depending on the signaling. It can > go from new to answered without progress. > > > > > On Sep 8, 2020, at 12:46 PM, Bilal Abbasi wrote: > > > > > > 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. > > > > > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From olle at zaark.com Fri Sep 11 06:13:34 2020 From: olle at zaark.com (Olle Frimanson) Date: Fri, 11 Sep 2020 08:13:34 +0200 Subject: [Freeswitch-users] Early media in the "wrong" direction In-Reply-To: <86FACA61-780F-4BA9-9115-E5728E70558B@freeswitch.org> References: <00de01d68123$84cd16c0$8e674440$@zaark.com> <86FACA61-780F-4BA9-9115-E5728E70558B@freeswitch.org> Message-ID: <002d01d68802$ae2dd640$0a8982c0$@zaark.com> Thanks Mike, Is this something you plan to support? BR/Olle Från: FreeSWITCH-users För Mike Jerris Skickat: den 9 september 2020 22:29 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Early media in the "wrong" direction Early media door phone like behavior like this is not currently supported. On Sep 2, 2020, at 8:20 AM, Olle Frimanson > wrote: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Fri Sep 11 21:45:02 2020 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 12 Sep 2020 02:45:02 +0500 Subject: [Freeswitch-users] [mod_amqp][golang] Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Fri Sep 11 22:09:42 2020 From: mike at freeswitch.org (Mike Jerris) Date: Fri, 11 Sep 2020 18:09:42 -0400 Subject: [Freeswitch-users] Early media in the "wrong" direction In-Reply-To: <002d01d68802$ae2dd640$0a8982c0$@zaark.com> References: <00de01d68123$84cd16c0$8e674440$@zaark.com> <86FACA61-780F-4BA9-9115-E5728E70558B@freeswitch.org> <002d01d68802$ae2dd640$0a8982c0$@zaark.com> Message-ID: Its not something we plan on adding anytime soon. > On Sep 11, 2020, at 2:13 AM, Olle Frimanson wrote: > > Thanks Mike, > > Is this something you plan to support? > > BR/Olle > > Från: FreeSWITCH-users > För Mike Jerris > Skickat: den 9 september 2020 22:29 > Till: FreeSWITCH Users Help > Ämne: Re: [Freeswitch-users] Early media in the "wrong" direction > > Early media door phone like behavior like this is not currently supported. > > >> On Sep 2, 2020, at 8:20 AM, Olle Frimanson > wrote: >> >> 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 >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Sep 11 23:03:51 2020 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 11 Sep 2020 20:03:51 -0300 Subject: [Freeswitch-users] [mod_amqp][golang] In-Reply-To: References: Message-ID: If it is of any help, I use https://github.com/vma/esl to connect from Golang to Freeswitch through ESL and it works very well. You can pretty much get anything from FreeSwitch using ESL. Regards, Guillermo On Fri, Sep 11, 2020 at 7:47 PM Bilal Abbasi wrote: > 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 > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Sat Sep 12 12:47:56 2020 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Sat, 12 Sep 2020 17:47:56 +0500 Subject: [Freeswitch-users] [mod_amqp][golang] In-Reply-To: References: Message-ID: Thanks Guillermo, I am using ESL since 2 years now for this system, but we have very high load of events, 10k+/second, and we miss some events at day end out of millions. To me it looks like we need some queue in middle to handle it. The reason i want to move to mod_amqp. Thanks again Abbasi On Sat, 12 Sep 2020 at 4:45 AM, Guillermo Ruiz Camauer wrote: > If it is of any help, I use https://github.com/vma/esl to connect from > Golang to Freeswitch through ESL and it works very well. > You can pretty much get anything from FreeSwitch using ESL. > > Regards, > > Guillermo > > On Fri, Sep 11, 2020 at 7:47 PM Bilal Abbasi wrote: > >> 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 >> >> >> >> _________________________________________________________________________ >> >> >> >> >> >> 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 at 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 at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> https://freeswitch.com > > > > -- > Guillermo Ruiz Camauer > > > _________________________________________________________________________ > > > > 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From obelousov at gmail.com Sat Sep 12 20:22:27 2020 From: obelousov at gmail.com (Oleg Belousov) Date: Sat, 12 Sep 2020 22:22:27 +0200 Subject: [Freeswitch-users] Add a=sendrecv onto sdp Message-ID: Hi. Please advise how I can add a=sendrecv parameter onto the SDP part of 183 Session progress. I'm using preAnswer in lua script to respond 183 with SDP to an invite. -- obelousov.tel -------------- next part -------------- An HTML attachment was scrubbed... URL: From th982a at googlemail.com Sat Sep 12 21:50:53 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 12 Sep 2020 23:50:53 +0200 Subject: [Freeswitch-users] issue with mod_python In-Reply-To: References: Message-ID: Hi David, Have you exported the pythonpath before you started freeswitch ? export PYTHONPATH=$PYTHONPATH:/usr/share/freeswitch/scripts/ should solve your problem, otherwise the import fails. best, Tamer On 2020-09-09 02:09, David Villasmil wrote: > 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": > 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 at gmail.com > > phone: +34669448337 > > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From shaun at sysconfig.cloud Mon Sep 14 09:10:16 2020 From: shaun at sysconfig.cloud (Shaun Stokes) Date: Mon, 14 Sep 2020 09:10:16 +0000 Subject: [Freeswitch-users] Add a=sendrecv onto sdp In-Reply-To: References: Message-ID: This might be what you're looking for, this is a SIP Profile parameter: https://freeswitch.org/confluence/display/FREESWITCH/NDLB It may also be possible to re-write the SDP per call: https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation#CodecNegotiation-RewritingSDP ________________________________ From: FreeSWITCH-users on behalf of Oleg Belousov Sent: 12 September 2020 22:22 To: FreeSWITCH Users Help Subject: [Freeswitch-users] Add a=sendrecv onto sdp Hi. Please advise how I can add a=sendrecv parameter onto the SDP part of 183 Session progress. I'm using preAnswer in lua script to respond 183 with SDP to an invite. -- obelousov.tel -------------- next part -------------- An HTML attachment was scrubbed... URL: From obelousov at gmail.com Mon Sep 14 17:13:27 2020 From: obelousov at gmail.com (Oleg Belousov) Date: Mon, 14 Sep 2020 19:13:27 +0200 Subject: [Freeswitch-users] Add a=sendrecv onto sdp In-Reply-To: References: Message-ID: Thank you for the advice, will check both options. -- obelousov.tel On Mon, Sep 14, 2020 at 11:28 AM Shaun Stokes wrote: > This might be what you're looking for, this is a SIP Profile parameter: > > https://freeswitch.org/confluence/display/FREESWITCH/NDLB > > It may also be possible to re-write the SDP per call: > > https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation#CodecNegotiation-RewritingSDP > > ------------------------------ > *From:* FreeSWITCH-users > on behalf of Oleg Belousov > *Sent:* 12 September 2020 22:22 > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Add a=sendrecv onto sdp > > Hi. > Please advise how I can add a=sendrecv parameter onto the SDP part of 183 > Session progress. I'm using preAnswer in lua script to respond 183 with SDP > to an invite. > -- > obelousov.tel > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Tue Sep 15 17:39:26 2020 From: dragos at freeswitch.org (Dragos Oancea) Date: Tue, 15 Sep 2020 20:39:26 +0300 Subject: [Freeswitch-users] [mod_amqp][golang] In-Reply-To: References: Message-ID: This one is good too: https://github.com/cgrates/fsock (ESL) On Sat, Sep 12, 2020 at 4:01 PM Bilal Abbasi wrote: > Thanks Guillermo, > I am using ESL since 2 years now for this system, but we have very high > load of events, 10k+/second, and we miss some events at day end out of > millions. To me it looks like we need some queue in middle to handle it. > The reason i want to move to mod_amqp. > Thanks again > > Abbasi > > On Sat, 12 Sep 2020 at 4:45 AM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> If it is of any help, I use https://github.com/vma/esl to connect from >> Golang to Freeswitch through ESL and it works very well. >> You can pretty much get anything from FreeSwitch using ESL. >> >> Regards, >> >> Guillermo >> >> On Fri, Sep 11, 2020 at 7:47 PM Bilal Abbasi wrote: >> >>> 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 >>> >>> >>> >>> _________________________________________________________________________ >>> >>> >>> >>> >>> >>> 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 at 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 at lists.freeswitch.org >>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> >>> https://freeswitch.com >> >> >> >> -- >> Guillermo Ruiz Camauer >> >> >> _________________________________________________________________________ >> >> >> >> 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Tue Sep 15 17:56:28 2020 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 15 Sep 2020 22:56:28 +0500 Subject: [Freeswitch-users] [mod_amqp][golang] In-Reply-To: References: Message-ID: Thanks Dragos, But i am looking for mod_amqp connector, now i have made that, i will publish it and add link here. Regards Abbasi On Tue, 15 Sep 2020 at 10:53 PM, Dragos Oancea wrote: > This one is good too: https://github.com/cgrates/fsock (ESL) > > On Sat, Sep 12, 2020 at 4:01 PM Bilal Abbasi wrote: > >> Thanks Guillermo, >> I am using ESL since 2 years now for this system, but we have very high >> load of events, 10k+/second, and we miss some events at day end out of >> millions. To me it looks like we need some queue in middle to handle it. >> The reason i want to move to mod_amqp. >> Thanks again >> >> Abbasi >> >> On Sat, 12 Sep 2020 at 4:45 AM, Guillermo Ruiz Camauer < >> grcamauer at gmail.com> wrote: >> >>> If it is of any help, I use https://github.com/vma/esl to connect from >>> Golang to Freeswitch through ESL and it works very well. >>> You can pretty much get anything from FreeSwitch using ESL. >>> >>> Regards, >>> >>> Guillermo >>> >>> On Fri, Sep 11, 2020 at 7:47 PM Bilal Abbasi >>> wrote: >>> >>>> 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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> >>>> >>>> >>>> >>>> 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 at 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 at lists.freeswitch.org >>>> >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> >>>> https://freeswitch.com >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> >>> _________________________________________________________________________ >>> >>> >>> >>> 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lylepratt at gmail.com Tue Sep 15 18:00:50 2020 From: lylepratt at gmail.com (Lyle Pratt) Date: Tue, 15 Sep 2020 13:00:50 -0500 Subject: [Freeswitch-users] Problem with the "file-only" Conference Layout Option Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Sep 15 20:42:40 2020 From: brian at freeswitch.com (Brian West) Date: Tue, 15 Sep 2020 15:42:40 -0500 Subject: [Freeswitch-users] Problem with the "file-only" Conference Layout Option In-Reply-To: References: Message-ID: Please report issues here https://github.com/signalwire/freeswitch/issues On Tue, Sep 15, 2020 at 1:34 PM Lyle Pratt wrote: > 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 > _________________________________________________________________________ > > 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Tue Sep 15 23:36:07 2020 From: davidswalkabout at gmail.com (David P) Date: Wed, 16 Sep 2020 11:36:07 +1200 Subject: [Freeswitch-users] One-way audio but not video on 10.4 In-Reply-To: References: Message-ID: Although we have STUN+TURN coturn (and FS) deployed in Oregon, a verto user in San Francisco today made a call in which only (private) host candidates were in the offer bundle SDP. They use IT-managed Ethernet, so any slowness in getting STUN is probably in their IT management of the connection...and not something we can do anything about unless our FS can support peer-reflexive candidates. They tell me their network has worked fine with Zoom, which is grating to hear. Is there any prospect of FS supporting peer-reflexive candidates? On Tue, Sep 1, 2020 at 12:22 AM < freeswitch-users-request at lists.freeswitch.org> wrote: > ---------- Forwarded message ---------- > From: David P > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Mon, 31 Aug 2020 08:36:05 +1200 > Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 > >> I strongly suspect the problem is due to FS choosing a different port > than is offered. I've seen this a few times by comparing the offer SDP > a=candidate entries with the address:port in FS log entries containing > 'Choose rtp'. This won't work if the user is behind a firewall that allows > incoming response only through ports opened for outgoing requests. > > Another thing I wonder about is when verto provides a BUNDLE offer, should > it use the same address and port for both audio and video. I think I've > seen that it doesn't, but maybe BUNDLE doesn't require it. > > --------- Forwarded message ---------- >> From: Nathan Stratton >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Sat, 29 Aug 2020 11:13:24 -0400 >> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >> Looks like your client is not hitting a stun server to get the public IP >> address, however, I have been wondering why it's not possible to make >> FreeSWITCH work without using STUN on the client as long as FreeSWITCH is >> on a public IP. >> >> ><> >> nathan stratton >> >> >> On Sat, Aug 29, 2020 at 3:00 AM David P >> wrote: >> >>> The client is current verto running on current Chrome/Win10. >>> >>> The client's SDP shows srflx and relay candidates from Twilio at same >>> network cost as the "private" host candidates, and FSv10.4 chooses the host >>> candidates. >>> >>> ---------- Forwarded message ---------- >>>> From: Mike Jerris >>>> >>>> If client still isn’t putting in non rfc1918 candidates in audio you >>>> need to figure out why that is first. >>>> >>>> On Aug 26, 2020, at 7:17 PM, David P wrote: >>>> >>>> I've introduced Twilio STUN and TURN to our iceServers since then. All >>>> have the same network cost in the SDP and FSv10.4 chooses a host. Verto >>>> reports the remote stream arrived but getStats shows zero audio bytes in or >>>> out, and it's silent to our user. I applied a display filter using our >>>> Freeswitch's public IP ip.src == 52.xx.yy.zz or ip.dst == ip.src == >>>> 52.xx.yy.zz and in addition to TLS I see some TCP ACKs from our user's >>>> 10.0.0.189 but I don't know what else to look for. >>>> >>>> >>>> ---------- Forwarded message ---------- >>>>> From: Mike Jerris >>>>> To: FreeSWITCH Users Help >>>>> Cc: >>>>> Bcc: >>>>> Date: Wed, 26 Aug 2020 16:19:56 -0400 >>>>> Subject: Re: [Freeswitch-users] One-way audio but not video on 10.4 >>>>> >>>> >>>> >>>>> Configure stun in your client so you get proper candidates that are >>>>> reachable. >>>>> >>>>> On Aug 19, 2020, at 6:18 PM, David P >>>>> wrote: >>>>> >>>>> Yesterday I had a verto call in which the