June 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problems migrating from 1.6.x to 1.8.x
by Laura Smith 08 Jun '19

08 Jun '19

Hi, I have a perfectly functioning FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit) instance running on Debian 8 (jessie). I'd like to move onwards and upwards, and so have installed FreeSWITCH Version 1.8.6-7-93b4c92e75~64bit (-7-93b4c92e75 64bit) on Debian 9 (stretch). In the hope that it would be as simple as copying accross /etc/freeswitch from old to new, I did that, followed by a reloadxml and, for good measure, a restart of the entire freeswitch instance. I have, of course, adjusted the relevant parameters in vars.xml in terms of hostname and IP addresses. However the new instance is refusing to allow my endpoints to register. Its sending 403 forbidden back. Any ideas ? Thanks ! Laura

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verto communicator
by Mansi Gupta 08 Jun '19

08 Jun '19

Hello everyone, I was testing verto communicator,it worked well. I have few queries. 1. Can we have layout like jitsi. 2. Is it possible to share/view files during conference. 3. In conference can we have feature like raise hand. Thank you -- Mansi Gupta

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conference video layers
by Madovsky 08 Jun '19

08 Jun '19

Hi folks, in mod_conference is it possible to for exemple select a 2 cam side by side but with let's say a zoom of the person taling more than 30 sec? Thanks Franck

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EVS Codec module for FreeSwitch ?
by Marcel Haldemann 07 Jun '19

07 Jun '19

Hi guys, Is there any change to get an EVS-Codec (https://www.iis.fraunhofer.de/en/ff/amm/communication/evs.html) implementation in FreeSwitch ? EVS is the successor of AMR-WB and As far as I understand it's royalty free for Network Devices (only mobile phone and table manufacturers got to license it, at least for now, and if it changes it can be centrally licensed at mpegla.com) Quote: "EVS Network Devices o Royalties waived during the initial term" from: https://www.mpegla.com/wp-content/uploads/EVSWeb.pdf General Info: https://www.mpegla.com/programs/evs/ Some info from other sources: https://www.businesswire.com/news/home/20170627006588/en/MPEG-LA-Introduces… regards, Marcel

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Re: [Freeswitch-users] iOS 12.3 / FreeSWITCH m-lines ordering issue
by David P 07 Jun '19

07 Jun '19

Hi Chad, When you've tested verto with Safari/iOS (say, v12), have you seen a call answered and then immediately hangup from the FS side? I get that repeatedly. Here's what I've tried: 1. We use the latest commercial release of FS, 20.19.4, on debian stretch 2. I merged your fix to jquery.FSRTC.js into ours, which has only minor changes from the tip of FS' repo 3. I tried our verto app on iOS 12 (using AWS Device Farm because I don't have Apple devices) 4. freeswitch.log shows verto.bye immediately after verto.answer, even in the same millisecond! I've attached a snippet of the log showing this. Cheers, David

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Issue | Shared Line
by jesse hiatt 06 Jun '19

06 Jun '19

We have multiple sites that use call groups with the same configuration however, only one site is experiencing any issues. I'm relatively new to FreeSWITCH, so please let me know if there is additional information I can provide. Thanks in advance for any advice. The sites are using FreeSWITCH version 1.2.14 The Polycom phones are using version 3.3.5 Here is the issue I see in the PCAP: [image: Screen Shot 2019-05-24 at 5.55.56 AM.png]

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Re: [Freeswitch-users] Inbound transferred calls are dropped
by Ramesh Kandasamy 05 Jun '19

05 Jun '19

There is a configuration under CuCM SIP trunk (or could be in SIP profile config) to avoid sending the hold reINVITE with 0.0.0.0 IP. You can try that out. With the log snippets, I think FS is expecting crypto keys for this SDP which CuCM will not provide as it is just a fake SDP. Thanks Ramesh On Wed, Jun 5, 2019 at 9:24 AM Stephen Barry via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Stephen Barry <stephen.barry(a)sensical.net> > To: "'freeswitch-users(a)lists.freeswitch.org'" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 5 Jun 2019 14:52:02 +0000 > Subject: Inbound transferred calls are dropped > Hi > > Please would anyone help point me in the right direction with a problem I > am having on FreeSWITCH (Version 1.8.5 -6-31281a0bf1 64bit) on FusionPBX > > I have FS connected to Cisco Call Manager over a SIP trunk (insecure) and > remote endpoints registering over the internet to FS secured with TLS and > SRTP > > <CUCM PBX> --- [SIP Trunk TCP + RTP] --- <FS> --- <FW/NAT> --- [TLS > Registration + SRTP] --- <FW/NAT> --- <Endpoint> > > I can make calls in all directions with audio, problem is when I try to do > an transfer from a phone on CUCM to the endpoint registered on FS the call > drops. > > Here is the capture https://pastebin.freeswitch.org/view/214d3b21 > > In this example it was a consultative transfer, the initial inbound call > was answered OK, but when the transfer was initiated the call dropped. > > My understanding of the issue from analysing this trace is when the call > is transferred CUCM sends a re-invite to put the call on hold maybe, that > causes the following problems: > > Line 1546 2019-06-05 15:33:30.169172 [WARNING] switch_core_media.c:4877 > RFC2543 from March 1999 called; They want their 0.0.0.0 hold method > back..... > Line 1587 2019-06-05 15:33:30.429173 [WARNING] switch_core_media.c:5551 > Crypto not negotiated but required. > Line 1590 2019-06-05 15:33:30.429173 [DEBUG] sofia.c:8216 Reinvite > resulted in codec negotiation failure. > > I have exhausted my limited knowledge of what is going wrong and how to > fix this and would be very grateful of assistance. > > Thanks > Steve > > > > > ---------- Forwarded message ---------- > From: Stephen Barry via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: "'freeswitch-users(a)lists.freeswitch.org'" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 05 Jun 2019 08:24:52 -0700 (PDT) > Subject: [Freeswitch-users] Inbound transferred calls are dropped > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

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Inbound transferred calls are dropped
by Stephen Barry 05 Jun '19

05 Jun '19

Hi Please would anyone help point me in the right direction with a problem I am having on FreeSWITCH (Version 1.8.5 -6-31281a0bf1 64bit) on FusionPBX I have FS connected to Cisco Call Manager over a SIP trunk (insecure) and remote endpoints registering over the internet to FS secured with TLS and SRTP <CUCM PBX> --- [SIP Trunk TCP + RTP] --- <FS> --- <FW/NAT> --- [TLS Registration + SRTP] --- <FW/NAT> --- <Endpoint> I can make calls in all directions with audio, problem is when I try to do an transfer from a phone on CUCM to the endpoint registered on FS the call drops. Here is the capture https://pastebin.freeswitch.org/view/214d3b21 In this example it was a consultative transfer, the initial inbound call was answered OK, but when the transfer was initiated the call dropped. My understanding of the issue from analysing this trace is when the call is transferred CUCM sends a re-invite to put the call on hold maybe, that causes the following problems: Line 1546 2019-06-05 15:33:30.169172 [WARNING] switch_core_media.c:4877 RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back..... Line 1587 2019-06-05 15:33:30.429173 [WARNING] switch_core_media.c:5551 Crypto not negotiated but required. Line 1590 2019-06-05 15:33:30.429173 [DEBUG] sofia.c:8216 Reinvite resulted in codec negotiation failure. I have exhausted my limited knowledge of what is going wrong and how to fix this and would be very grateful of assistance. Thanks Steve

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having problems configuring source of master
by John Covici 01 Jun '19

01 Jun '19

Hi. I am having a problem configuring freeswitch master. I have libvpx-1.7.0 but I get the following complaint when running ./configure checking for v8-6.1_static >= 6.1.298... checking for v8fs_static >= 6.1.298... checking for v8 >= 6.1.298... configure: error: You need to either install libv8-6.1-dev (>= 6.1.298), libv8fs-dev (>= 6.1.298) or disable mod_v8 in modules.conf Since I have 7.0 should not this work? Where is it looking for these files? I do have a pc file for this, but its called vpx.pc and had the information for libvpx package, so what am I missing? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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Re: [Freeswitch-users] Run script that calls exts and plays recording
by Vallimamod Abdullah 01 Jun '19

01 Jun '19

Hi, There is no b-leg when the call is answered by freeswitch. You need to create new sessions for your subsequent calls in your lua script. Best Regards, -- Vallimamod Abdullah SIP Solutions vma(a)sip.solutions linkedin.com/in/vallimamod . > On 1 Jun 2019, at 13:54, Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: Bipin Patel <bipin(a)xbipin.com <mailto:bipin@xbipin.com>> > Subject: Re: [Freeswitch-users] Run script that calls exts and plays recording > Date: 1 June 2019 at 12:34:19 CEST > To: <freeswitch-users(a)lists.freeswitch.org> > > > Hi, > > I have a requirement where one phone calls the number 55 which fs answers and asks user to record his message and hangup when complete which works fine but after he hangs up I want to run a lua script which can take this recorded message and call multiple phones and play it to them but this part doesn't work even when using api_hangup_hook because fs replies not session available. > > Is there any way to keep the b leg which is fs itself answering active and run the script? It's fs answering and recording the message so not able to use other commands related to bridge like continue on fail or hang up after bridge or transfer etc. > > Any pointers will be helpful > > > > Regards, > > Bipin > > On June 1, 2019 1:51:09 PM Bipin Patel via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> wrote: > >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <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 <https://signalwire.community/> > > Professional FreeSWITCH Services > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > https://freeswitch.com <https://freeswitch.com/> > > Official FreeSWITCH Sites > https://freeswitch.com/oss <https://freeswitch.com/oss> > https://freeswitch.org/confluence <https://freeswitch.org/confluence> > https://cluecon.com <https://cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > https://freeswitch.com <https://freeswitch.com/>

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