From borik.internet at gmail.com Sun Dec 1 16:18:07 2019 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Sun, 1 Dec 2019 19:18:07 +0300 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Hello, Bilal! I have found, that mod_avmd successful identifies beeps above ~700Hz. I don't see events for beeps at 450-700Hz, and in sources lowest frequency limited to 450Hz. ... i can achieve this via wait_for_silence... this - such? пт, 29 нояб. 2019 г. в 13:31, Bilal Abbasi : > Hi, > Thanks for your reply, is there any way i can achieve this via > wait_for_silence, i am just wondering if some one has done this in > freeswitch or even this is something possible. > > Regards > Abbasi > > On Fri, 29 Nov 2019 at 4:54 AM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> From GitHub: >> >> Currently, in limited testing, we are able to get satisfactory results in >> determining what is a human and what is a machine, but *there is much >> more to do*: >> >> - Emit events when a decision is made (Not sure, Machine, or Human). >> <== This is a TODO item >> - Make sure that we are unlocking and cleaning up where necessary. >> >> The last commit was 4 years ago. It looks like this is an abandoned >> project. >> >> Guillermo >> >> On Thu, Nov 28, 2019 at 1:01 PM Bilal Abbasi wrote: >> >>> Hi users, >>> I am using mod_avmd for quite a while, but i now want to use amd like >>> its in asterisk. >>> I found a related module here >>> https://github.com/seanbright/mod_amd >>> >>> I am not able to successfully generate events through that, do someone >>> has a working dialplan example along with parameters. That would be >>> really helpful. >>> >>> P.S: i have tested on freeswitch 1.10, any special version for this. >>> >>> 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 -- With best regards Dmitry Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sun Dec 1 19:34:03 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 1 Dec 2019 20:34:03 +0100 Subject: [Freeswitch-users] call transfer In-Reply-To: References: Message-ID: do this, to redirect: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+redirect -giovanni On Sat, Nov 30, 2019 at 1:15 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Iif you're just forwarding to another destination, you'd better let the > destination answer the call, instead of you answering _after_ bridging it: > > <-- As > Ciprian pointed out, using the IP is not always a good idea. Are you also > using ACL? > > > data="{...}sofia/external/${destination_number}@$${OTHER_PROVIDER}"/> > <--- shouldn't be here, > you're answering the call after you've transferred it. > > > > You're also not using a gateway, but an IP address, instead create a > gateway and use it like so: > > data="{...}sofia/gateway/[your-gateway]/${destination_number}"/> > > About the media, according to Brian: > > bypass_media=false >> > > Does nothing. > > >> proxy_media=false >> > > Does nothing. > > But that was in regards to t.38, so... > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Fri, Nov 29, 2019 at 11:03 PM Ciprian Dosoftei < > ciprian.dosoftei at gmail.com> wrote: > >> 1- is this the right way? >>> >> >> It really depends on your circumstances and the use case, so it's >> difficult to tell. Also, the network address criteria may not be as stable >> as expected, perhaps another variable is better suited for that condition. >> >> >>> 2- when i do like this, do RTP packages go over my server? >>> >> >> In most circumstances, the RTP will flow through your server indeed. >> >> >>> 3- what should i do, just to transfer signalling, not to transfer media >>> over my server? >>> >> >> Check bypass_media and/or bypass_media_after_bridge: >> https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#VariablesMasterList-bypass_media >> _________________________________________________________________________ >> >> 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From abelcubano at gmail.com Mon Dec 2 01:32:33 2019 From: abelcubano at gmail.com (Abel Monzon) Date: Sun, 1 Dec 2019 20:32:33 -0500 Subject: [Freeswitch-users] bash script with variables In-Reply-To: References: <95CD35A0-EC6A-4F76-B28E-4C3D19BA073B@freeswitch.org> Message-ID: what is the dialplan command to run /root/scripts/test1.sh? El jue., 28 nov. 2019 a las 8:15, Dragos Oancea () escribió: > Maybe it has something to do with mutt config and maybe mutt is even > clearing an env variable with the same name when it's getting executed as > user 'freeswitch' from the dialplan. When you execute that script from > command line you do it as a regular user, try to become the freeswitch user > (su) and see if it still works. > > > On Wed, Nov 27, 2019 at 8:31 PM Mike Jerris wrote: > >> Do you not see the file at all? Might be a permissions or working >> directory issue. >> >> > On Nov 27, 2019, at 8:54 AM, thomas peterseil < >> thomas.peterseil at mine-project.eu> wrote: >> > >> > hello freeswitch-list, >> > i would like to start from the dialplan a bash script and the >> > freeswitch is handing over one variable to the script. in the cli i >> > see the correct start of the script: >> > >> > [NOTICE] mod_dptools.c:2120 Executing command: /root/scripts/test1.sh >> 3333 >> > >> > the script is very simple: >> > >> > #!/bin/bash >> > number=$1 >> > >> > mutt -s "Call $number" thomas.peterseil at mine-project.eu < >> > /root/mailtexte/registrierung.txt >> > echo "$number" > test1.txt >> > >> > i get the email with the variable in the subject, that works fine, but >> > i can´t see the 3333 in the test1.txt file. >> > when i start the script from the command line with ./test1.sh 3333 all >> > is working fine. can somebody give me a hint why it doesn´t work from >> > the dialplan. >> >> >> _________________________________________________________________________ >> >> 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 linmin at qza.io Tue Dec 3 17:42:39 2019 From: linmin at qza.io (LinMin) Date: Wed, 4 Dec 2019 03:42:39 +1000 Subject: [Freeswitch-users] Outbound calls to remote gateway on dynamic IP Message-ID: Hi folks, I have a Freeswitch server with a PSTN gateway logged in as a normal user, which gives access to the gateway on a dynamic IP address (trying to avoid static IP and extra DNS settings in this situation). Calls coming in from the gateway to FS work fine, but trying to route calls to the gateway hit a snag, because instead of sending calls to: @gateway_IP , they go to: @gateway_IP^ which results in the gateway attempting to dial it's own user ID via the PSTN line. I've tried using this option in the dialplan extension to work around it: which results in the above situation. Manually inserting the remote IP works: .... however, I'm hoping there's a way that I can fish and insert the IP address on the fly, or otherwise tell the gateway to dial the correct number. Cheers, Francis From moshe.rosenberg at gmail.com Tue Dec 3 02:50:59 2019 From: moshe.rosenberg at gmail.com (Moshe Rosenberg) Date: Mon, 2 Dec 2019 21:50:59 -0500 Subject: [Freeswitch-users] Recording RECORD_APPEND=true Prefix Message-ID: Hi right now i had send my dial-plan to RECORD_APPEND=true and its working fine how ever it appends the recording to the end of the file, i would really like to append the recording to the begging of the file is there away to do this on freeswitch ? -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data phone.cloud Moshe at dataphone.cloud -------------- next part -------------- An HTML attachment was scrubbed... URL: From roman at dissauer.net Tue Dec 3 21:24:32 2019 From: roman at dissauer.net (Roman Dissauer) Date: Tue, 3 Dec 2019 22:24:32 +0100 Subject: [Freeswitch-users] t38_gateway hangs up non-deterministic with normal clearing Message-ID: we have an issue with t38_gateway where calls which should be transcoded from PCMA to t38 get hangup non-deterministic. It doesn’t matter if there is only one call or if there are hundrets of calls on the freeswitch. Sometimes it happens that we get several faxes transcoded but then we again have several faxes which don’t go through. problem exists (tested) on FreeSWITCH 1.6 and FreeSWITCH 1.10. this is the dialplan section where t38_gateway is set up for b-leg on an inbound call: ... tested freeswitch versions: FreeSWITCH Version 1.6.20-37-987c9b9~64bit (-37-987c9b9 64bit) FreeSWITCH Version 1.10.1-release-12-f9990221e6~64bit (-release-12-f9990221e6 64bit) -> version where logfile is created this is the debug section where the call gets hung up: 287b2821-05a1-4ff6-9449-dac913b803eb 2019-12-03 20:32:23.257123 [DEBUG] switch_core_media.c:5285 sofia/internal/46wp809758 at 10.23.101.10 T38 IS POSSIBLE on request cc222460-f51e-453e-9a09-c2e97e6b8af9 2019-12-03 20:32:23.259118 [ALERT] switch_rtp.c:1595 sofia/external/+4318033811 at 193.32.51.146:5060 audio stat 100.00 673/673 flaws: 0 mos: 4.50 v: 0.00 0.00/0.00 cc222460-f51e-453e-9a09-c2e97e6b8af9 2019-12-03 20:32:23.259118 [DEBUG] switch_ivr_bridge.c:824 sofia/internal/46wp809758 at 10.23.101.10 ending bridge by request from write function cc222460-f51e-453e-9a09-c2e97e6b8af9 2019-12-03 20:32:23.259118 [DEBUG] switch_ivr_bridge.c:916 BRIDGE THREAD DONE [sofia/external/+4318033811 at 193.32.51.146:5060] 287b2821-05a1-4ff6-9449-dac913b803eb 2019-12-03 20:32:23.260120 [ALERT] switch_ivr_bridge.c:1893 sofia/internal/46wp809758 at 10.23.101.10 receive message [UNBRIDGE] full logfile of call is enclosed fsctl debug settings: fsctl loglevel debug fsctl debug_level 9 -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: 20191203_faxproblem_1.txt URL: From brian at freeswitch.com Tue Dec 3 23:06:10 2019 From: brian at freeswitch.com (Brian West) Date: Tue, 3 Dec 2019 17:06:10 -0600 Subject: [Freeswitch-users] Recording RECORD_APPEND=true Prefix In-Reply-To: References: Message-ID: Not possible. /b On Tue, Dec 3, 2019 at 1:55 PM Moshe Rosenberg wrote: > Hi > > right now i had send my dial-plan to RECORD_APPEND=true and its working > fine how ever it appends the recording to the end of the file, i would > really like to append the recording to the begging of the file > > is there away to do this on freeswitch ? > > -- > Moshe Rosenberg > Tel. 718 633 1444 > Cel. 347 678 3993 > www.data phone.cloud > Moshe at dataphone.cloud > > > _________________________________________________________________________ > > 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 mike at freeswitch.org Tue Dec 3 23:52:41 2019 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 3 Dec 2019 15:52:41 -0800 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: Message-ID: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. > On Nov 29, 2019, at 11:28 AM, John Covici wrote: > > I have the log of the call which looks normal. My guess is that rtp > is not properly being sent out, for some reason. The hangup cause is > always normal_clearing. > > On Fri, 29 Nov 2019 13:06:12 -0500, > David Villasmil wrote: >> >> [1 ] >> [1.1 ] >> Do you have any trace? >> >> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: >> >>> Some more information -- even after pressing a digit and getting >>> audio, it hangs up after about 30 seconds. >>> >>> On Fri, 29 Nov 2019 10:46:21 -0500, >>> John Covici wrote: >>>> >>>> Hi. I finally was able to upgrade fs to master as of llast night. >>>> Its working well, except if I use portaudio to make a call. This all >>>> worked find in fs 1.6.20. >>>> >>>> When I call someone I cannot hear anything until I send it a dtmf >>>> (rfc2283) and then things work normally, at least I can hear >>>> something. I had a look at the logs, but nothing strange in there >>>> after typing the digit. >>>> >>>> Also, I cannot call a local extension from port audio, even though the >>>> extension is registered and can be called from another extension. It >>>> immediately goes to voicemail. >>>> >>>> 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 >>>> >>> >>> -- >>> 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 >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> [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 > > _________________________________________________________________________ > > 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 mike at freeswitch.org Tue Dec 3 23:53:49 2019 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 3 Dec 2019 15:53:49 -0800 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: <5ACAED7A-2011-41A0-A481-007E565CCEE1@freeswitch.org> Its possible the issue is whats being filtered by the codec in use. Do you only see this on some codecs but not g.711? > On Dec 1, 2019, at 8:18 AM, Dmitriy Borisov wrote: > > Hello, Bilal! > > I have found, that mod_avmd successful identifies beeps above ~700Hz. I don't see events for beeps at 450-700Hz, and in sources lowest frequency limited to 450Hz. > > ... i can achieve this via wait_for_silence... > > this - such? > > пт, 29 нояб. 2019 г. в 13:31, Bilal Abbasi >: > Hi, > Thanks for your reply, is there any way i can achieve this via wait_for_silence, i am just wondering if some one has done this in freeswitch or even this is something possible. > > Regards > Abbasi > > On Fri, 29 Nov 2019 at 4:54 AM, Guillermo Ruiz Camauer > wrote: > From GitHub: > > Currently, in limited testing, we are able to get satisfactory results in determining what is a human and what is a machine, but there is much more to do: > > Emit events when a decision is made (Not sure, Machine, or Human). <== This is a TODO item > Make sure that we are unlocking and cleaning up where necessary. > The last commit was 4 years ago. It looks like this is an abandoned project. > > Guillermo > > On Thu, Nov 28, 2019 at 1:01 PM Bilal Abbasi > wrote: > Hi users, > I am using mod_avmd for quite a while, but i now want to use amd like its in asterisk. > I found a related module here > https://github.com/seanbright/mod_amd > I am not able to successfully generate events through that, do someone has a working dialplan example along with parameters. That would be really helpful. > > P.S: i have tested on freeswitch 1.10, any special version for this. > > 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 > > -- > With best regards > Dmitry Borisov > _________________________________________________________________________ > > 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 covici at ccs.covici.com Wed Dec 4 03:39:33 2019 From: covici at ccs.covici.com (John Covici) Date: Tue, 03 Dec 2019 22:39:33 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Message-ID: Thanks for your response. I don't know how to read the pcap file, so here is a link to the file, maybe you can figure it out better than I can. https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb On Tue, 03 Dec 2019 18:52:41 -0500, Mike Jerris wrote: > > [1 ] > [1.1 ] > Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. > > > On Nov 29, 2019, at 11:28 AM, John Covici wrote: > > > > I have the log of the call which looks normal. My guess is that rtp > > is not properly being sent out, for some reason. The hangup cause is > > always normal_clearing. > > > > On Fri, 29 Nov 2019 13:06:12 -0500, > > David Villasmil wrote: > >> > >> [1 ] > >> [1.1 ] > >> Do you have any trace? > >> > >> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > >> > >>> Some more information -- even after pressing a digit and getting > >>> audio, it hangs up after about 30 seconds. > >>> > >>> On Fri, 29 Nov 2019 10:46:21 -0500, > >>> John Covici wrote: > >>>> > >>>> Hi. I finally was able to upgrade fs to master as of llast night. > >>>> Its working well, except if I use portaudio to make a call. This all > >>>> worked find in fs 1.6.20. > >>>> > >>>> When I call someone I cannot hear anything until I send it a dtmf > >>>> (rfc2283) and then things work normally, at least I can hear > >>>> something. I had a look at the logs, but nothing strange in there > >>>> after typing the digit. > >>>> > >>>> Also, I cannot call a local extension from port audio, even though the > >>>> extension is registered and can be called from another extension. It > >>>> immediately goes to voicemail. > >>>> > >>>> 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 > >>>> > >>> > >>> -- > >>> 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 > >> > >> -- > >> Regards, > >> > >> David Villasmil > >> email: david.villasmil.work at gmail.com > >> phone: +34669448337 > >> [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 > > > > _________________________________________________________________________ > > > > 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 > [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 mike at freeswitch.org Wed Dec 4 05:14:02 2019 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 3 Dec 2019 21:14:02 -0800 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Message-ID: <4B78721F-3BFE-41CC-B984-D01896459B2B@freeswitch.org> Needs the sip in the pcap too to understand what the rtp is. this looks like capture started after call was already going. > On Dec 3, 2019, at 7:39 PM, John Covici wrote: > > Thanks for your response. > > I don't know how to read the pcap file, so here is a link to the > file, maybe you can figure it out better than I can. > > https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb > > > On Tue, 03 Dec 2019 18:52:41 -0500, > Mike Jerris wrote: >> >> [1 ] >> [1.1 ] >> Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. >> >>> On Nov 29, 2019, at 11:28 AM, John Covici wrote: >>> >>> I have the log of the call which looks normal. My guess is that rtp >>> is not properly being sent out, for some reason. The hangup cause is >>> always normal_clearing. >>> >>> On Fri, 29 Nov 2019 13:06:12 -0500, >>> David Villasmil wrote: >>>> >>>> [1 ] >>>> [1.1 ] >>>> Do you have any trace? >>>> >>>> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: >>>> >>>>> Some more information -- even after pressing a digit and getting >>>>> audio, it hangs up after about 30 seconds. >>>>> >>>>> On Fri, 29 Nov 2019 10:46:21 -0500, >>>>> John Covici wrote: >>>>>> >>>>>> Hi. I finally was able to upgrade fs to master as of llast night. >>>>>> Its working well, except if I use portaudio to make a call. This all >>>>>> worked find in fs 1.6.20. >>>>>> >>>>>> When I call someone I cannot hear anything until I send it a dtmf >>>>>> (rfc2283) and then things work normally, at least I can hear >>>>>> something. I had a look at the logs, but nothing strange in there >>>>>> after typing the digit. >>>>>> >>>>>> Also, I cannot call a local extension from port audio, even though the >>>>>> extension is registered and can be called from another extension. It >>>>>> immediately goes to voicemail. >>>>>> >>>>>> Thanks in advance for any suggestions. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rajil.s at gmail.com Wed Dec 4 05:52:33 2019 From: rajil.s at gmail.com (Rajil Saraswat) Date: Wed, 4 Dec 2019 11:22:33 +0530 Subject: [Freeswitch-users] Audiocode MP-114 FXS_FXO device setup Message-ID: Hello, My trusted Linksys SPA3102 died and I am trying to replace it with an Audiocodes MP-114 device which has 2 FXS ports and 2 FXO ports. My first plan was to get the FXS port working. I specified my FreeSwitch PBX details in VOIP>SIP Definitions>Proxy & Registration and specified my number in VOIP>GW and IP to IP>Hunt Group>Endpoint Phone Manager. With this i was able to successfully register an extension '204' on the pbx. However, if i make a call from a softphone to this number the calls do not come through, and the ATA logs says 4d:11h:17m:15s ( lgr_TrnkGrp)(9672 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 204 4d:11h:17m:15s ( lgr_psbrdif)(9673 ) !! [ERROR] MotherBoard::GetEndPoint- Can't find EndPoint for Dest:204 Source:mod_sofia SourceIp:ac10020a 4d:11h:17m:15s ( lgr_call)(9674 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 204 Any idea where do i define the 'Endpoint' for number 204? Thanks, Rajil The .ini file for ATA is as follows ;************** ;** Ini File ** ;************** ;Board: MP-114 FXS_FXO ;Board Type: 56 ;Serial Number: 4929340 ;Slot Number: 1 ;Software Version: 6.60A.355.004 ;DSP Software Version: 204IM3=> 660.15 ;Board IP Address: 172.16.2.4 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 172.16.2.1 ;Ram size: 32M Flash size: 8M ;Num of DSP Cores: 1 Num DSP Channels: 4 ;Profile: NONE ;License Key limits aren't active full features capabilities are available !; ;---------------------------------------------- [SYSTEM Params] SyslogServerIP = 10.1.1.89 ;NTPServerIP_abs is hidden but has non-default value NTPServerUTCOffset = 19800 ;VpFileLastUpdateTime is hidden but has non-default value NTPServerIP = '172.16.2.1' [BSP Params] PCMLawSelect = 3 RoutingTableHopsCountColumn = 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 [Analog Params] PolarityReversalType = 1 MinFlashHookTime = 100 [ControlProtocols Params] AdminStateLockControl = 0 [MGCP Params] [MEGACO Params] EP_Num_0 = 0 EP_Num_1 = 1 EP_Num_2 = 1 EP_Num_3 = 0 EP_Num_4 = 0 [Voice Engine Params] CallProgressTonesFilename = 'usa_tones_13.dat' FaxTransportMode = 0 V22ModemTransportType = 0 V23ModemTransportType = 0 V32ModemTransportType = 0 V34ModemTransportType = 0 RFC2833TxPayloadType = 101 [WEB Params] LogoWidth = '208' HTTPSCipherString = 'RC4:EXP' [SIP Params] MAXDIGITS = 3 ISREGISTERNEEDED = 1 AUTHENTICATIONMODE = 0 GWDEBUGLEVEL = 5 ;ISPRACKREQUIRED is hidden but has non-default value ENABLEEARLYMEDIA = 1 PROXYNAME = 'BLR' REGISTRARIP = '172.16.2.10' SIPGATEWAYNAME = 'pbx.blah.com' USERNAME = '204' ;PASSWORD is hidden but has non-default value ;SHOULDREGISTER is hidden but has non-default value SUBSCRIPTIONMODE = 0 REGISTRARNAME = 'pbx.blah.com' REGISTRARTRANSPORTTYPE = 0 MSLDAPPRIMARYKEY = 'telephoneNumber' [IPsec Params] [SNMP Params] [ DspTemplates ] ; ; *** TABLE DspTemplates *** ; This table contains hidden elements and will not be exposed. ; This table exists on board and will be saved during restarts. ; [ \DspTemplates ] [ PREFIX ] FORMAT PREFIX_Index = PREFIX_DestinationPrefix, PREFIX_DestAddress, PREFIX_SourcePrefix, PREFIX_ProfileId, PREFIX_MeteringCode, PREFIX_DestPort, PREFIX_SrcIPGroupID, PREFIX_DestHostPrefix, PREFIX_DestIPGroupID, PREFIX_SrcHostPrefix, PREFIX_TransportType, PREFIX_SrcTrunkGroupID, PREFIX_DestSRD, PREFIX_CostGroup, PREFIX_ForkingGroup, PREFIX_CallSetupRulesSetId; PREFIX 0 = "10", "10.1.10.10", "*", 0, 255, 0, -1, "", -1, "", -1, -1, -1, "", -1, -1; PREFIX 1 = "20", "10.1.10.11", "*", 0, 255, 0, -1, "", -1, "", -1, -1, -1, "", -1, -1; [ \PREFIX ] [ TrunkGroup ] FORMAT TrunkGroup_Index = TrunkGroup_TrunkGroupNum, TrunkGroup_FirstTrunkId, TrunkGroup_FirstBChannel, TrunkGroup_LastBChannel, TrunkGroup_FirstPhoneNumber, TrunkGroup_ProfileId, TrunkGroup_LastTrunkId, TrunkGroup_Module; TrunkGroup 0 = 1, 255, 1, 1, "204", 1, 255, 255; [ \TrunkGroup ] [ ProxyIp ] FORMAT ProxyIp_Index = ProxyIp_IpAddress, ProxyIp_TransportType, ProxyIp_ProxySetId; ProxyIp 0 = "172.16.2.10", 0, 0; [ \ProxyIp ] [ TxDtmfOption ] FORMAT TxDtmfOption_Index = TxDtmfOption_Type; TxDtmfOption 0 = 4; [ \TxDtmfOption ] [ TrunkGroupSettings ] FORMAT TrunkGroupSettings_Index = TrunkGroupSettings_TrunkGroupId, TrunkGroupSettings_ChannelSelectMode, TrunkGroupSettings_RegistrationMode, TrunkGroupSettings_GatewayName, TrunkGroupSettings_ContactUser, TrunkGroupSettings_ServingIPGroup, TrunkGroupSettings_DedicatedConnectionMode, TrunkGroupSettings_MWIInterrogationType, TrunkGroupSettings_TrunkGroupName; TrunkGroupSettings 0 = 1, 0, 0, "", "", -1, 0, 255, ""; [ \TrunkGroupSettings ] [ EnableCallerId ] FORMAT EnableCallerId_Index = EnableCallerId_IsEnabled, EnableCallerId_Port, EnableCallerId_PortType; EnableCallerId 0 = 1, 1, "FXS"; [ \EnableCallerId ] [ ProxySet ] FORMAT ProxySet_Index = ProxySet_EnableProxyKeepAlive, ProxySet_ProxyKeepAliveTime, ProxySet_ProxyLoadBalancingMethod, ProxySet_IsProxyHotSwap, ProxySet_SRD, ProxySet_ClassificationInput, ProxySet_ProxyRedundancyMode, ProxySet_KeepAliveFailureResp, ProxySet_HomingSuccessDetectionRetries; ProxySet 0 = 0, 60, 0, 0, 0, 0, -1, "", 1; [ \ProxySet ] [ CodersGroup0 ] FORMAT CodersGroup0_Index = CodersGroup0_Name, CodersGroup0_pTime, CodersGroup0_rate, CodersGroup0_PayloadType, CodersGroup0_Sce; CodersGroup0 0 = "g711Alaw64k", 20, 255, -1, 0; CodersGroup0 1 = "g711Ulaw64k", 20, 255, -1, 0; [ \CodersGroup0 ] [ RoutingRuleGroups ] FORMAT RoutingRuleGroups_Index = RoutingRuleGroups_LCREnable, RoutingRuleGroups_LCRAverageCallLength, RoutingRuleGroups_LCRDefaultCost; RoutingRuleGroups 0 = 0, 1, 1; [ \RoutingRuleGroups ] [ ResourcePriorityNetworkDomains ] FORMAT ResourcePriorityNetworkDomains_Index = ResourcePriorityNetworkDomains_Name, ResourcePriorityNetworkDomains_Ip2TelInterworking; ResourcePriorityNetworkDomains 1 = "dsn", 1; ResourcePriorityNetworkDomains 2 = "dod", 1; ResourcePriorityNetworkDomains 3 = "drsn", 1; ResourcePriorityNetworkDomains 5 = "uc", 1; ResourcePriorityNetworkDomains 7 = "cuc", 1; [ \ResourcePriorityNetworkDomains ] -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Wed Dec 4 06:45:21 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 4 Dec 2019 11:45:21 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: <5ACAED7A-2011-41A0-A481-007E565CCEE1@freeswitch.org> References: <5ACAED7A-2011-41A0-A481-007E565CCEE1@freeswitch.org> Message-ID: Hi, Actually i don't want to make decision on beep(normally we get beep after 10,11 seconds for call being answered). I want to categories call as human/voicemail in initial 3-5 seconds. The reason i am trying to chose something similar to asterisk’s amd. If someone can help me achieve that via any available functions/modules(like i said might be done via wait for silence) Regards Abbasi On Wed, 4 Dec 2019 at 5:32 AM, Mike Jerris wrote: > Its possible the issue is whats being filtered by the codec in use. Do > you only see this on some codecs but not g.711? > > > On Dec 1, 2019, at 8:18 AM, Dmitriy Borisov > wrote: > > Hello, Bilal! > > I have found, that mod_avmd successful identifies beeps above ~700Hz. I > don't see events for beeps at 450-700Hz, and in sources lowest > frequency limited to 450Hz. > > ... i can achieve this via wait_for_silence... > > > this - such? > > пт, 29 нояб. 2019 г. в 13:31, Bilal Abbasi : > >> Hi, >> Thanks for your reply, is there any way i can achieve this via >> wait_for_silence, i am just wondering if some one has done this in >> freeswitch or even this is something possible. >> >> Regards >> Abbasi >> >> On Fri, 29 Nov 2019 at 4:54 AM, Guillermo Ruiz Camauer < >> grcamauer at gmail.com> wrote: >> >>> From GitHub: >>> >>> Currently, in limited testing, we are able to get satisfactory results >>> in determining what is a human and what is a machine, but *there is >>> much more to do*: >>> >>> - Emit events when a decision is made (Not sure, Machine, or >>> Human). <== This is a TODO item >>> - Make sure that we are unlocking and cleaning up where necessary. >>> >>> The last commit was 4 years ago. It looks like this is an abandoned >>> project. >>> >>> Guillermo >>> >>> On Thu, Nov 28, 2019 at 1:01 PM Bilal Abbasi >>> wrote: >>> >>>> Hi users, >>>> I am using mod_avmd for quite a while, but i now want to use amd like >>>> its in asterisk. >>>> I found a related module here >>>> https://github.com/seanbright/mod_amd >>>> >>>> I am not able to successfully generate events through that, do someone >>>> has a working dialplan example along with parameters. That would be >>>> really helpful. >>>> >>>> P.S: i have tested on freeswitch 1.10, any special version for this. >>>> >>>> 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 > > > > -- > With best regards > Dmitry Borisov > _________________________________________________________________________ > > 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 covici at ccs.covici.com Wed Dec 4 12:07:57 2019 From: covici at ccs.covici.com (John Covici) Date: Wed, 04 Dec 2019 07:07:57 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: <4B78721F-3BFE-41CC-B984-D01896459B2B@freeswitch.org> References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> <4B78721F-3BFE-41CC-B984-D01896459B2B@freeswitch.org> Message-ID: OK, I have updated the file including portrange 5060-5080 as well as the rtp portrange. The same link should still work. Thanks. On Wed, 04 Dec 2019 00:14:02 -0500, Mike Jerris wrote: > > [1 ] > [1.1 ] > Needs the sip in the pcap too to understand what the rtp is. this looks like capture started after call was already going. > > > On Dec 3, 2019, at 7:39 PM, John Covici wrote: > > > > Thanks for your response. > > > > I don't know how to read the pcap file, so here is a link to the > > file, maybe you can figure it out better than I can. > > > > https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb > > > > > > On Tue, 03 Dec 2019 18:52:41 -0500, > > Mike Jerris wrote: > >> > >> [1 ] > >> [1.1 ] > >> Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. > >> > >>> On Nov 29, 2019, at 11:28 AM, John Covici wrote: > >>> > >>> I have the log of the call which looks normal. My guess is that rtp > >>> is not properly being sent out, for some reason. The hangup cause is > >>> always normal_clearing. > >>> > >>> On Fri, 29 Nov 2019 13:06:12 -0500, > >>> David Villasmil wrote: > >>>> > >>>> [1 ] > >>>> [1.1 ] > >>>> Do you have any trace? > >>>> > >>>> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > >>>> > >>>>> Some more information -- even after pressing a digit and getting > >>>>> audio, it hangs up after about 30 seconds. > >>>>> > >>>>> On Fri, 29 Nov 2019 10:46:21 -0500, > >>>>> John Covici wrote: > >>>>>> > >>>>>> Hi. I finally was able to upgrade fs to master as of llast night. > >>>>>> Its working well, except if I use portaudio to make a call. This all > >>>>>> worked find in fs 1.6.20. > >>>>>> > >>>>>> When I call someone I cannot hear anything until I send it a dtmf > >>>>>> (rfc2283) and then things work normally, at least I can hear > >>>>>> something. I had a look at the logs, but nothing strange in there > >>>>>> after typing the digit. > >>>>>> > >>>>>> Also, I cannot call a local extension from port audio, even though the > >>>>>> extension is registered and can be called from another extension. It > >>>>>> immediately goes to voicemail. > >>>>>> > >>>>>> Thanks in advance for any suggestions. > > [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 covici at ccs.covici.com Wed Dec 4 12:08:31 2019 From: covici at ccs.covici.com (John Covici) Date: Wed, 04 Dec 2019 07:08:31 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: <4B78721F-3BFE-41CC-B984-D01896459B2B@freeswitch.org> References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> <4B78721F-3BFE-41CC-B984-D01896459B2B@freeswitch.org> Message-ID: the call is *398300 so you can find it. On Wed, 04 Dec 2019 00:14:02 -0500, Mike Jerris wrote: > > [1 ] > [1.1 ] > Needs the sip in the pcap too to understand what the rtp is. this looks like capture started after call was already going. > > > On Dec 3, 2019, at 7:39 PM, John Covici wrote: > > > > Thanks for your response. > > > > I don't know how to read the pcap file, so here is a link to the > > file, maybe you can figure it out better than I can. > > > > https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb > > > > > > On Tue, 03 Dec 2019 18:52:41 -0500, > > Mike Jerris wrote: > >> > >> [1 ] > >> [1.1 ] > >> Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. > >> > >>> On Nov 29, 2019, at 11:28 AM, John Covici wrote: > >>> > >>> I have the log of the call which looks normal. My guess is that rtp > >>> is not properly being sent out, for some reason. The hangup cause is > >>> always normal_clearing. > >>> > >>> On Fri, 29 Nov 2019 13:06:12 -0500, > >>> David Villasmil wrote: > >>>> > >>>> [1 ] > >>>> [1.1 ] > >>>> Do you have any trace? > >>>> > >>>> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > >>>> > >>>>> Some more information -- even after pressing a digit and getting > >>>>> audio, it hangs up after about 30 seconds. > >>>>> > >>>>> On Fri, 29 Nov 2019 10:46:21 -0500, > >>>>> John Covici wrote: > >>>>>> > >>>>>> Hi. I finally was able to upgrade fs to master as of llast night. > >>>>>> Its working well, except if I use portaudio to make a call. This all > >>>>>> worked find in fs 1.6.20. > >>>>>> > >>>>>> When I call someone I cannot hear anything until I send it a dtmf > >>>>>> (rfc2283) and then things work normally, at least I can hear > >>>>>> something. I had a look at the logs, but nothing strange in there > >>>>>> after typing the digit. > >>>>>> > >>>>>> Also, I cannot call a local extension from port audio, even though the > >>>>>> extension is registered and can be called from another extension. It > >>>>>> immediately goes to voicemail. > >>>>>> > >>>>>> Thanks in advance for any suggestions. > > [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 gmaruzz at gmail.com Wed Dec 4 13:02:55 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 4 Dec 2019 14:02:55 +0100 Subject: [Freeswitch-users] t38_gateway hangs up non-deterministic with normal clearing In-Reply-To: References: Message-ID: On Tue, Dec 3, 2019 at 11:19 PM Roman Dissauer wrote: > we have an issue with t38_gateway where calls which should be transcoded > from PCMA to t38 get hangup non-deterministic. It doesn’t matter if there > is only one call or if there are hundrets of calls on the freeswitch. > Sometimes it happens that we get several faxes transcoded but then we again > have several faxes which don’t go through. problem exists (tested) on > FreeSWITCH 1.6 and FreeSWITCH 1.10. > > this is the dialplan section where t38_gateway is set up for b-leg on an > inbound call: > > > > break="never"/> > expression="^(43732601458)(\d{0,4})$“> > ... > > data="fax_enable_t38=true"> > data="nolocal:sip_execute_on_image=t38_gateway self nocng"> > data="{sip_cid_type=pid,sip_contact_user=accountcode,absolute_codec_string='PCMA at 20i > ,PCMU at 20i,G729'}sofia/internal/accountcode at 10.23.101.10^+${ > dialed_extension}@10.23.101.10"> > ... > > > > > Would you detail, line by line, what is the intended purpose of each line of dialplan? Also, your topology, and the intended result -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Dec 4 14:02:43 2019 From: brian at freeswitch.com (Brian West) Date: Wed, 4 Dec 2019 08:02:43 -0600 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: You do know we sell a commercial AMD module? On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi wrote: > Hi users, > I am using mod_avmd for quite a while, but i now want to use amd like its > in asterisk. > I found a related module here > https://github.com/seanbright/mod_amd > > I am not able to successfully generate events through that, do someone has > a working dialplan example along with parameters. That would be > really helpful. > > P.S: i have tested on freeswitch 1.10, any special version for this. > > 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 -- 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 shahnawaj.khan1990 at gmail.com Wed Dec 4 16:28:38 2019 From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan) Date: Wed, 4 Dec 2019 21:58:38 +0530 Subject: [Freeswitch-users] Unable to compile mod_v8 Message-ID: Hi, I need to test freeswitch asr sample mod_v8 is required to run the javascript. It is not there by default, So i tried to compile it but facing the issue as. ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or directory #include Please let me know if anyone have faced the same issue and its resolution. Thanks & Regards, shahnawaj From asilva at wirelessmundi.com Wed Dec 4 17:13:10 2019 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 4 Dec 2019 18:13:10 +0100 Subject: mod_watson available for debian 10 Message-ID: Hi, Is mod_watson (https://freeswitch.org/confluence/display/FREESWITCH/mod_watson) available for debian buster? Can it be activated on compile version like g729? Thanks. -- Saludos / Regards / Cumprimentos António Silva From bilaln018 at gmail.com Wed Dec 4 17:51:39 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Wed, 4 Dec 2019 22:51:39 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Yes Brian, My client is looking for something similar to that, but he does not have budget to spend around 50$/channel. Dont get me wrong, we love FreeSWITCH, but being the dev factory we have to understand client's needs.(and those gets very strong, when someone says we need whats freely available in Asteriks ;-) ) On Wed, Dec 4, 2019 at 7:33 PM Brian West wrote: > You do know we sell a commercial AMD module? > > On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi wrote: > >> Hi users, >> I am using mod_avmd for quite a while, but i now want to use amd like its >> in asterisk. >> I found a related module here >> https://github.com/seanbright/mod_amd >> >> I am not able to successfully generate events through that, do someone >> has a working dialplan example along with parameters. That would be >> really helpful. >> >> P.S: i have tested on freeswitch 1.10, any special version for this. >> >> 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 > > > > -- > > 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] > _________________________________________________________________________ > > 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 sdisanti.eteleco at gmail.com Wed Dec 4 20:09:59 2019 From: sdisanti.eteleco at gmail.com (Sean DiSanti) Date: Wed, 4 Dec 2019 12:09:59 -0800 Subject: [Freeswitch-users] Unable to compile mod_v8 In-Reply-To: References: Message-ID: You most likely need to uncomment the line to load the v8 module in /etc/freeswitch/autoload_configs/modules.conf.xml You're going to run into a few other issues getting that pizza demo working (speaking from experience) depending on where you got the code for SpeechTools.jm. I don't remember all of them as it's been about a year since I went through that one, but you'll almost definitely need to add use('XML') to the top of your SpeechTools.jm On Wed, Dec 4, 2019 at 8:29 AM Shahnawaj Khan wrote: > Hi, > > I need to test freeswitch asr sample > > > > expression="true"> > > > > mod_v8 is required to run the javascript. It is not there by default, > So i tried to compile it but facing the issue as. > > ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or > directory > #include > > Please let me know if anyone have faced the same issue and its resolution. > > Thanks & Regards, > shahnawaj > > _________________________________________________________________________ > > 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 piotr at dataandsignal.com Thu Dec 5 02:00:53 2019 From: piotr at dataandsignal.com (Piotr Gregor) Date: Thu, 5 Dec 2019 02:00:53 +0000 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Module AVMD is optimized for frequencies above 440 Hz. There is limit hardcoded: /*! Minimum beep frequency in Hertz */ #define AVMD_MIN_FREQUENCY (440.0) You can simply change that, and recompile. BTW: You know voicemails beeping at such low frequencies? 😉 Wow, it's not beep anymore, it's boop. Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Wed, Dec 4, 2019 at 6:25 PM Bilal Abbasi wrote: > Yes Brian, > My client is looking for something similar to that, but he does not have > budget to spend around 50$/channel. Dont get me wrong, we love FreeSWITCH, > but being the dev factory we have to understand client's needs.(and those > gets very strong, when someone says we need whats freely available in > Asteriks ;-) ) > > On Wed, Dec 4, 2019 at 7:33 PM Brian West wrote: > >> You do know we sell a commercial AMD module? >> >> On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi >> wrote: >> >>> Hi users, >>> I am using mod_avmd for quite a while, but i now want to use amd like >>> its in asterisk. >>> I found a related module here >>> https://github.com/seanbright/mod_amd >>> >>> I am not able to successfully generate events through that, do someone >>> has a working dialplan example along with parameters. That would be >>> really helpful. >>> >>> P.S: i have tested on freeswitch 1.10, any special version for this. >>> >>> 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 >> >> >> >> -- >> >> 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] >> _________________________________________________________________________ >> >> 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 ciprian.dosoftei at gmail.com Thu Dec 5 04:15:34 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Wed, 4 Dec 2019 23:15:34 -0500 Subject: [Freeswitch-users] Unable to compile mod_v8 In-Reply-To: References: Message-ID: Shahnawaj -- Make sure you have libv8-dev installed on the machine you're building the module (and libv8 on the machines where you use the module). -C On Wed, Dec 4, 2019, 4:05 PM Sean DiSanti wrote: > You most likely need to uncomment the line to load the v8 module in > /etc/freeswitch/autoload_configs/modules.conf.xml You're going to run into > a few other issues getting that pizza demo working (speaking from > experience) depending on where you got the code for SpeechTools.jm. I don't > remember all of them as it's been about a year since I went through that > one, but you'll almost definitely need to add use('XML') to the top of your > SpeechTools.jm > > On Wed, Dec 4, 2019 at 8:29 AM Shahnawaj Khan < > shahnawaj.khan1990 at gmail.com> wrote: > >> Hi, >> >> I need to test freeswitch asr sample >> >> >> >> > expression="true"> >> >> >> >> mod_v8 is required to run the javascript. It is not there by default, >> So i tried to compile it but facing the issue as. >> >> ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or >> directory >> #include >> >> Please let me know if anyone have faced the same issue and its resolution. >> >> Thanks & Regards, >> shahnawaj >> >> _________________________________________________________________________ >> >> 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 ciprian.dosoftei at gmail.com Thu Dec 5 04:24:22 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Wed, 4 Dec 2019 23:24:22 -0500 Subject: [Freeswitch-users] Recording RECORD_APPEND=true Prefix In-Reply-To: References: Message-ID: Moshe -- Use some sox magic, after the calls complete. This also offers a good opportunity to decouple the moving parts of your application and run this kind of postprocessing on other machines, allowing your FreeSWITCH nodes to focus on what they do best (real time comms). -C On Tue, Dec 3, 2019, 6:23 PM Brian West wrote: > Not possible. > > /b > > > On Tue, Dec 3, 2019 at 1:55 PM Moshe Rosenberg > wrote: > >> Hi >> >> right now i had send my dial-plan to RECORD_APPEND=true and its working >> fine how ever it appends the recording to the end of the file, i would >> really like to append the recording to the begging of the file >> >> is there away to do this on freeswitch ? >> >> -- >> Moshe Rosenberg >> Tel. 718 633 1444 >> Cel. 347 678 3993 >> www.data phone.cloud >> Moshe at dataphone.cloud >> >> >> _________________________________________________________________________ >> >> 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] > _________________________________________________________________________ > > 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 shahnawaj.khan1990 at gmail.com Thu Dec 5 04:42:02 2019 From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan) Date: Thu, 5 Dec 2019 10:12:02 +0530 Subject: [Freeswitch-users] Unable to compile mod_v8 In-Reply-To: References: Message-ID: Hi Ciprian, I am using CentOs 7 and installed v8-devel.x86_64 1:3.14.5.10-25.el7 on my machine. Now it is giving list of errors such as In file included from mod_v8.h:36:0, from mod_v8.cpp:68: ./include/javascript.hpp:163:53: error: ‘FunctionCallbackInfo’ in namespace ‘v8’ does not name a type typedef void_pointer_t (*ConstructorCallback)(const v8::FunctionCallbackInfo& info); ^ ./include/javascript.hpp:163:57: error: ISO C++ forbids declaration of ‘parameter’ with no type [-fpermissive] typedef void_pointer_t (*ConstructorCallback)(const v8::FunctionCallbackInfo& info); ^ ./include/javascript.hpp:163:77: error: expected ‘,’ or ‘...’ before ‘<’ token typedef void_pointer_t (*ConstructorCallback)(const v8::FunctionCallbackInfo& info); ^ ./include/javascript.hpp:168:2: error: ‘FunctionCallback’ in namespace ‘v8’ does not name a type v8::FunctionCallback func; /* Function callback */ ^ ./include/javascript.hpp:174:2: error: ‘AccessorGetterCallback’ in namespace ‘v8’ does not name a type v8::AccessorGetterCallback get; /* The property getter */ is this due to some version issue of v8. Thanks & Regards, shahnawaj On Thu, Dec 5, 2019 at 9:53 AM Ciprian Dosoftei wrote: > > Shahnawaj -- > > Make sure you have libv8-dev installed on the machine you're building the module (and libv8 on the machines where you use the module). > > -C > > On Wed, Dec 4, 2019, 4:05 PM Sean DiSanti wrote: >> >> You most likely need to uncomment the line to load the v8 module in /etc/freeswitch/autoload_configs/modules.conf.xml You're going to run into a few other issues getting that pizza demo working (speaking from experience) depending on where you got the code for SpeechTools.jm. I don't remember all of them as it's been about a year since I went through that one, but you'll almost definitely need to add use('XML') to the top of your SpeechTools.jm >> >> On Wed, Dec 4, 2019 at 8:29 AM Shahnawaj Khan wrote: >>> >>> Hi, >>> >>> I need to test freeswitch asr sample >>> >>> >>> >>> >>> >>> >>> >>> mod_v8 is required to run the javascript. It is not there by default, >>> So i tried to compile it but facing the issue as. >>> >>> ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or directory >>> #include >>> >>> Please let me know if anyone have faced the same issue and its resolution. >>> >>> Thanks & Regards, >>> shahnawaj >>> >>> _________________________________________________________________________ >>> >>> 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 From yehavi.bourvine at gmail.com Thu Dec 5 05:40:22 2019 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 5 Dec 2019 07:40:22 +0200 Subject: [Freeswitch-users] Unable to compile mod_v8 In-Reply-To: References: Message-ID: I had the same issue on Centos; after a few hours or attepmts I decided to re-write my JS scripts to LUA... __Yehavi: ‫בתאריך יום ה׳, 5 בדצמ׳ 2019 ב-6:42 מאת ‪Shahnawaj Khan‬‏ <‪ shahnawaj.khan1990 at gmail.com‬‏>:‬ > Hi Ciprian, > > I am using CentOs 7 and installed v8-devel.x86_64 1:3.14.5.10-25.el7 > on my machine. Now it is giving list of errors such as > > In file included from mod_v8.h:36:0, > from mod_v8.cpp:68: > ./include/javascript.hpp:163:53: error: ‘FunctionCallbackInfo’ in > namespace ‘v8’ does not name a type > typedef void_pointer_t (*ConstructorCallback)(const > v8::FunctionCallbackInfo& info); > ^ > ./include/javascript.hpp:163:57: error: ISO C++ forbids declaration of > ‘parameter’ with no type [-fpermissive] > typedef void_pointer_t (*ConstructorCallback)(const > v8::FunctionCallbackInfo& info); > ^ > ./include/javascript.hpp:163:77: error: expected ‘,’ or ‘...’ before ‘<’ > token > typedef void_pointer_t (*ConstructorCallback)(const > v8::FunctionCallbackInfo& info); > > ^ > ./include/javascript.hpp:168:2: error: ‘FunctionCallback’ in namespace > ‘v8’ does not name a type > v8::FunctionCallback func; /* Function callback */ > ^ > ./include/javascript.hpp:174:2: error: ‘AccessorGetterCallback’ in > namespace ‘v8’ does not name a type > v8::AccessorGetterCallback get; /* The property getter */ > > is this due to some version issue of v8. > > Thanks & Regards, > shahnawaj > > On Thu, Dec 5, 2019 at 9:53 AM Ciprian Dosoftei > wrote: > > > > Shahnawaj -- > > > > Make sure you have libv8-dev installed on the machine you're building > the module (and libv8 on the machines where you use the module). > > > > -C > > > > On Wed, Dec 4, 2019, 4:05 PM Sean DiSanti > wrote: > >> > >> You most likely need to uncomment the line to load the v8 module in > /etc/freeswitch/autoload_configs/modules.conf.xml You're going to run into > a few other issues getting that pizza demo working (speaking from > experience) depending on where you got the code for SpeechTools.jm. I don't > remember all of them as it's been about a year since I went through that > one, but you'll almost definitely need to add use('XML') to the top of your > SpeechTools.jm > >> > >> On Wed, Dec 4, 2019 at 8:29 AM Shahnawaj Khan < > shahnawaj.khan1990 at gmail.com> wrote: > >>> > >>> Hi, > >>> > >>> I need to test freeswitch asr sample > >>> > >>> > >>> > >>> expression="true"> > >>> > >>> > >>> > >>> mod_v8 is required to run the javascript. It is not there by default, > >>> So i tried to compile it but facing the issue as. > >>> > >>> ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or > directory > >>> #include > >>> > >>> Please let me know if anyone have faced the same issue and its > resolution. > >>> > >>> Thanks & Regards, > >>> shahnawaj > >>> > >>> > _________________________________________________________________________ > >>> > >>> 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 > > _________________________________________________________________________ > > 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 shahnawaj.khan1990 at gmail.com Thu Dec 5 07:20:02 2019 From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan) Date: Thu, 5 Dec 2019 12:50:02 +0530 Subject: [Freeswitch-users] Unable to compile mod_v8 In-Reply-To: References: Message-ID: Hi Yehavi, Is there any chance that I can get that sample lua script. Thanks & Regards, Shahnawaj On Thu, Dec 5, 2019 at 12:14 PM Yehavi Bourvine wrote: > > I had the same issue on Centos; after a few hours or attepmts I decided to re-write my JS scripts to LUA... > > __Yehavi: > > ‫בתאריך יום ה׳, 5 בדצמ׳ 2019 ב-6:42 מאת ‪Shahnawaj Khan‬‏ <‪shahnawaj.khan1990 at gmail.com‬‏>:‬ >> >> Hi Ciprian, >> >> I am using CentOs 7 and installed v8-devel.x86_64 1:3.14.5.10-25.el7 >> on my machine. Now it is giving list of errors such as >> >> In file included from mod_v8.h:36:0, >> from mod_v8.cpp:68: >> ./include/javascript.hpp:163:53: error: ‘FunctionCallbackInfo’ in >> namespace ‘v8’ does not name a type >> typedef void_pointer_t (*ConstructorCallback)(const >> v8::FunctionCallbackInfo& info); >> ^ >> ./include/javascript.hpp:163:57: error: ISO C++ forbids declaration of >> ‘parameter’ with no type [-fpermissive] >> typedef void_pointer_t (*ConstructorCallback)(const >> v8::FunctionCallbackInfo& info); >> ^ >> ./include/javascript.hpp:163:77: error: expected ‘,’ or ‘...’ before ‘<’ token >> typedef void_pointer_t (*ConstructorCallback)(const >> v8::FunctionCallbackInfo& info); >> ^ >> ./include/javascript.hpp:168:2: error: ‘FunctionCallback’ in namespace >> ‘v8’ does not name a type >> v8::FunctionCallback func; /* Function callback */ >> ^ >> ./include/javascript.hpp:174:2: error: ‘AccessorGetterCallback’ in >> namespace ‘v8’ does not name a type >> v8::AccessorGetterCallback get; /* The property getter */ >> >> is this due to some version issue of v8. >> >> Thanks & Regards, >> shahnawaj >> >> On Thu, Dec 5, 2019 at 9:53 AM Ciprian Dosoftei >> wrote: >> > >> > Shahnawaj -- >> > >> > Make sure you have libv8-dev installed on the machine you're building the module (and libv8 on the machines where you use the module). >> > >> > -C >> > >> > On Wed, Dec 4, 2019, 4:05 PM Sean DiSanti wrote: >> >> >> >> You most likely need to uncomment the line to load the v8 module in /etc/freeswitch/autoload_configs/modules.conf.xml You're going to run into a few other issues getting that pizza demo working (speaking from experience) depending on where you got the code for SpeechTools.jm. I don't remember all of them as it's been about a year since I went through that one, but you'll almost definitely need to add use('XML') to the top of your SpeechTools.jm >> >> >> >> On Wed, Dec 4, 2019 at 8:29 AM Shahnawaj Khan wrote: >> >>> >> >>> Hi, >> >>> >> >>> I need to test freeswitch asr sample >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> mod_v8 is required to run the javascript. It is not there by default, >> >>> So i tried to compile it but facing the issue as. >> >>> >> >>> ./include/javascript.hpp:35:16: fatal error: v8.h: No such file or directory >> >>> #include >> >>> >> >>> Please let me know if anyone have faced the same issue and its resolution. >> >>> >> >>> Thanks & Regards, >> >>> shahnawaj >> >>> >> >>> _________________________________________________________________________ >> >>> >> >>> 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 >> >> _________________________________________________________________________ >> >> 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 From Jaan.Kaja at enghouse.com Thu Dec 5 10:38:47 2019 From: Jaan.Kaja at enghouse.com (Jaan Kaja) Date: Thu, 5 Dec 2019 10:38:47 +0000 Subject: [Freeswitch-users] Long increasing audio delay Message-ID: Hello, We are testing webrtc via mod_verto with the verto_communicator demo. After some initial troubles with certificates, we have it running from Chrome and Firefox. Very nice! But, when we leave the connection on for some time, the audio in the direction from the browser to a phone becomes increasingly longer. We have clocked ~11 seconds, if we leave it on long enough. The audio in the other direction has a normal IP telephony delay. We have not found any browser dependencies. The FreeSWITCH version is 1.10.1 (latest stable release). The phone end has been a Blink softphone, or a trunk from a Cisco UCM. The latter negotiated mulaw PCM with a 20 ms packet size. We have tried different server OS:es: * Windows Server 2019 gives the fastest buildup of delay. * Debian installed in a Windows Substrate for Linux gives a slower buildup of delay, but it does continuously increase. * CentOS virtualized in VMWare doesn't have any buildup of delay, and the delay is a normal IP telephony delay. * So, the more Windows, the worse... I have a Wireshark trace, and I can get a TLS stream from it. I can install a key file to decode the stream, but the configuration wants a protocol, and I don't know what to enter there. There are no dissectors for Websocket. So I'm at a bit of a loss, as to how to debug this. I don't see anything obvious in the freeswitch.log. Questions: * Has anyone used webrtc on Windows? * Has anyone had the same problem? * What can be done to debug the problem? Turn on some logging? We could package a Linux machine to our customers, but as everything else is on Windows, it would be nice to make it work. Best regards, Jaan -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Thu Dec 5 20:14:02 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Thu, 5 Dec 2019 15:14:02 -0500 Subject: [Freeswitch-users] Long increasing audio delay In-Reply-To: References: Message-ID: The long story short is: the more you separate a real-time application from the metal, the worse it gets. WSL does implement most (if not all) Linux kernel syscalls but ultimately translates into NT kernel specific implementations which will inherently induce some additional delay. KVM/Xen are reasonably good in terms of providing good timers for real-time applications, but even then, the difference compared against metal is noticeable, however, less of an issue for reasonable volume as well as other circumstances (primarily, if trans-coding is involved). Debian on metal is your best choice, though it should not be a showstopper over Xen or KVM if your circumstances are not too demanding. On Thu, 5 Dec 2019 at 14:35, Jaan Kaja wrote: > Hello, > > > > We are testing webrtc via mod_verto with the verto_communicator demo. > After some initial troubles with certificates, we have it running from > Chrome and Firefox. Very nice! > > > > But, when we leave the connection on for some time, the audio in the > direction from the browser to a phone becomes increasingly longer. We have > clocked ~11 seconds, if we leave it on long enough. The audio in the other > direction has a normal IP telephony delay. We have not found any browser > dependencies. > > > > The FreeSWITCH version is 1.10.1 (latest stable release). The phone end > has been a Blink softphone, or a trunk from a Cisco UCM. The latter > negotiated mulaw PCM with a 20 ms packet size. > > > > We have tried different server OS:es: > > · Windows Server 2019 gives the fastest buildup of delay. > > · Debian installed in a Windows Substrate for Linux gives a > slower buildup of delay, but it does continuously increase. > > · CentOS virtualized in VMWare doesn’t have any buildup of delay, > and the delay is a normal IP telephony delay. > > · So, the more Windows, the worse… > > > > I have a Wireshark trace, and I can get a TLS stream from it. I can > install a key file to decode the stream, but the configuration wants a > protocol, and I don’t know what to enter there. There are no dissectors for > Websocket. So I’m at a bit of a loss, as to how to debug this. I don’t see > anything obvious in the freeswitch.log. > > > > Questions: > > · Has anyone used webrtc on Windows? > > · Has anyone had the same problem? > > · What can be done to debug the problem? Turn on some logging? > > > > We could package a Linux machine to our customers, but as everything else > is on Windows, it would be nice to make it work. > > > > Best regards, > > Jaan > _________________________________________________________________________ > > 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 -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Thu Dec 5 21:05:02 2019 From: gregor at infomedia.si (Gregor Nanger) Date: Thu, 5 Dec 2019 22:05:02 +0100 Subject: [Freeswitch-users] Long increasing audio delay In-Reply-To: References: Message-ID: Hi! We are using Windows on production for years, also with verto. As I remember, we had similar problems. I think we solved with timer settings in verto.conf.xml to: Try to set it and see if it works. Best regards, Gregor V V čet., 5. dec. 2019 ob 20:10 je oseba Jaan Kaja napisala: > Hello, > > > > We are testing webrtc via mod_verto with the verto_communicator demo. > After some initial troubles with certificates, we have it running from > Chrome and Firefox. Very nice! > > > > But, when we leave the connection on for some time, the audio in the > direction from the browser to a phone becomes increasingly longer. We have > clocked ~11 seconds, if we leave it on long enough. The audio in the other > direction has a normal IP telephony delay. We have not found any browser > dependencies. > > > > The FreeSWITCH version is 1.10.1 (latest stable release). The phone end > has been a Blink softphone, or a trunk from a Cisco UCM. The latter > negotiated mulaw PCM with a 20 ms packet size. > > > > We have tried different server OS:es: > > · Windows Server 2019 gives the fastest buildup of delay. > > · Debian installed in a Windows Substrate for Linux gives a > slower buildup of delay, but it does continuously increase. > > · CentOS virtualized in VMWare doesn’t have any buildup of delay, > and the delay is a normal IP telephony delay. > > · So, the more Windows, the worse… > > > > I have a Wireshark trace, and I can get a TLS stream from it. I can > install a key file to decode the stream, but the configuration wants a > protocol, and I don’t know what to enter there. There are no dissectors for > Websocket. So I’m at a bit of a loss, as to how to debug this. I don’t see > anything obvious in the freeswitch.log. > > > > Questions: > > · Has anyone used webrtc on Windows? > > · Has anyone had the same problem? > > · What can be done to debug the problem? Turn on some logging? > > > > We could package a Linux machine to our customers, but as everything else > is on Windows, it would be nice to make it work. > > > > Best regards, > > Jaan > _________________________________________________________________________ > > 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 -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From roman at dissauer.net Fri Dec 6 00:23:23 2019 From: roman at dissauer.net (Roman Dissauer) Date: Fri, 6 Dec 2019 01:23:23 +0100 Subject: [Freeswitch-users] t38_gateway hangs up non-deterministic with normal clearing In-Reply-To: References: Message-ID: <255A1D15-BA6C-4D2C-BFE1-BB8B49567ABB@dissauer.net> this is our topology: customer pbx <-> kamailio + rtpengine <-> several FreeSWITCH Machines <-> kamailio <-> several interconnection carriers what we have to address for inbound/outbound calls: inbound call comes from any interconnection carrier which mostly does NOT have t.38 faxing what is no issue because we have reliable internet lines to our carriers. Our customers want to have t.38 faxing because of reliability of faxing over standard internet lines. So we have to transcode g.711a codec to t.38 in freeswitch. Same is true for outbound calls where customer pbxes send t.38 re-Invite but the upstream interconnection carrier does not accept t.38. sometimes faxing is fine with this mechanism but mostly the fax is rejected by a „NORMAL CLEARING“ as soon as the re-Invite is coming in. FreeSWITCH is sending BYE to both call legs. what we intend to do with our dialplan on an example - incoming calls 1. call is coming in via external profile with g.711a Codec 2. set refuse_t38=true sets t38 to be rejected coming from a-leg - external carrier 3. export fax_enable_t38=true will enable t38 both call legs 4. export nolocal:sip_execute_on_image=t38_image=t38_gateway self nocng will enable t38 only on b-leg - customer side 5. call is being bridged to kamailio where it gets routed to the customer thank you! best regards, Roman > Am 04.12.2019 um 14:02 schrieb Giovanni Maruzzelli : > > On Tue, Dec 3, 2019 at 11:19 PM Roman Dissauer > wrote: > we have an issue with t38_gateway where calls which should be transcoded from PCMA to t38 get hangup non-deterministic. It doesn’t matter if there is only one call or if there are hundrets of calls on the freeswitch. Sometimes it happens that we get several faxes transcoded but then we again have several faxes which don’t go through. problem exists (tested) on FreeSWITCH 1.6 and FreeSWITCH 1.10. > > this is the dialplan section where t38_gateway is set up for b-leg on an inbound call: > > > > > > > > > ... > > > > > > > Would you detail, line by line, what is the intended purpose of each line of dialplan? > > Also, your topology, and the intended result > > -giovanni > > _________________________________________________________________________ > > 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 gmaruzz at gmail.com Fri Dec 6 06:56:08 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 6 Dec 2019 07:56:08 +0100 Subject: [Freeswitch-users] t38_gateway hangs up non-deterministic with normal clearing In-Reply-To: <255A1D15-BA6C-4D2C-BFE1-BB8B49567ABB@dissauer.net> References: <255A1D15-BA6C-4D2C-BFE1-BB8B49567ABB@dissauer.net> Message-ID: Just a couple thoughts : - fax over G711 VoIP is totally unreliable, hit or miss, random. It's a fact of nature, no matter how good your internet lines are - maybe you want try to answer the incoming, wait for 1 second so RTP flow is well established, and then bridge -giovanni On Fri, Dec 6, 2019 at 1:57 AM Roman Dissauer wrote: > this is our topology: > > customer pbx <-> kamailio + rtpengine <-> several FreeSWITCH Machines <-> > kamailio <-> several interconnection carriers > > what we have to address for inbound/outbound calls: > inbound call comes from any interconnection carrier which mostly does NOT > have t.38 faxing what is no issue because we have reliable internet lines > to our carriers. Our customers want to have t.38 faxing because of > reliability of faxing over standard internet lines. So we have to transcode > g.711a codec to t.38 in freeswitch. Same is true for outbound calls where > customer pbxes send t.38 re-Invite but the upstream interconnection carrier > does not accept t.38. > > sometimes faxing is fine with this mechanism but mostly the fax is > rejected by a „NORMAL CLEARING“ as soon as the re-Invite is coming in. > FreeSWITCH is sending BYE to both call legs. > > what we intend to do with our dialplan on an example - incoming calls > > 1. call is coming in via external profile with g.711a Codec > 2. set refuse_t38=true sets t38 to be rejected coming from a-leg - > external carrier > 3. export fax_enable_t38=true will enable t38 both call legs > 4. export nolocal:sip_execute_on_image=t38_image=t38_gateway self nocng > will enable t38 only on b-leg - customer side > 5. call is being bridged to kamailio where it gets routed to the customer > > thank you! > > best regards, > Roman > > Am 04.12.2019 um 14:02 schrieb Giovanni Maruzzelli : > > On Tue, Dec 3, 2019 at 11:19 PM Roman Dissauer wrote: > >> we have an issue with t38_gateway where calls which should be transcoded >> from PCMA to t38 get hangup non-deterministic. It doesn’t matter if there >> is only one call or if there are hundrets of calls on the freeswitch. >> Sometimes it happens that we get several faxes transcoded but then we again >> have several faxes which don’t go through. problem exists (tested) on >> FreeSWITCH 1.6 and FreeSWITCH 1.10. >> >> this is the dialplan section where t38_gateway is set up for b-leg on an >> inbound call: >> >> >> >> > break="never"/> >> > expression="^(43732601458)(\d{0,4})$“> >> ... >> >> > data="fax_enable_t38=true"> >> > data="nolocal:sip_execute_on_image=t38_gateway self nocng"> >> > data="{sip_cid_type=pid,sip_contact_user=accountcode,absolute_codec_string='PCMA at 20i >> ,PCMU at 20i,G729'}sofia/internal/accountcode at 10.23.101.10^+${ >> dialed_extension}@10.23.101.10"> >> ... >> >> >> >> >> > > Would you detail, line by line, what is the intended purpose of each line > of dialplan? > > Also, your topology, and the intended result > > -giovanni > > _________________________________________________________________________ > > 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Jaan.Kaja at enghouse.com Fri Dec 6 12:04:33 2019 From: Jaan.Kaja at enghouse.com (Jaan Kaja) Date: Fri, 6 Dec 2019 12:04:33 +0000 Subject: [Freeswitch-users] Long increasing audio delay In-Reply-To: References: Message-ID: <39aa9728dd444801bf08f084cd24a07f@enghouse.com> Many thanks, Gregor! It now works like a charm on my bare metal workstation. Audio comes up fast after call setup, and the audio delay buildup is gone. How did you figure it out? Regarding virtualization, a Windows 2016 Server under Hyper-V still has the problem, but it works fine under VMWare. It may be that people have done more performance tweaking on the VMWare system. /Jaan From: FreeSWITCH-users On Behalf Of Gregor Nanger Sent: den 5 december 2019 22:05 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Long increasing audio delay Hi! We are using Windows on production for years, also with verto. As I remember, we had similar problems. I think we solved with timer settings in verto.conf.xml to: Try to set it and see if it works. Best regards, Gregor V V čet., 5. dec. 2019 ob 20:10 je oseba Jaan Kaja > napisala: Hello, We are testing webrtc via mod_verto with the verto_communicator demo. After some initial troubles with certificates, we have it running from Chrome and Firefox. Very nice! But, when we leave the connection on for some time, the audio in the direction from the browser to a phone becomes increasingly longer. We have clocked ~11 seconds, if we leave it on long enough. The audio in the other direction has a normal IP telephony delay. We have not found any browser dependencies. The FreeSWITCH version is 1.10.1 (latest stable release). The phone end has been a Blink softphone, or a trunk from a Cisco UCM. The latter negotiated mulaw PCM with a 20 ms packet size. We have tried different server OS:es: • Windows Server 2019 gives the fastest buildup of delay. • Debian installed in a Windows Substrate for Linux gives a slower buildup of delay, but it does continuously increase. • CentOS virtualized in VMWare doesn’t have any buildup of delay, and the delay is a normal IP telephony delay. • So, the more Windows, the worse… I have a Wireshark trace, and I can get a TLS stream from it. I can install a key file to decode the stream, but the configuration wants a protocol, and I don’t know what to enter there. There are no dissectors for Websocket. So I’m at a bit of a loss, as to how to debug this. I don’t see anything obvious in the freeswitch.log. Questions: • Has anyone used webrtc on Windows? • Has anyone had the same problem? • What can be done to debug the problem? Turn on some logging? We could package a Linux machine to our customers, but as everything else is on Windows, it would be nice to make it work. Best regards, Jaan _________________________________________________________________________ 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 -- Gregor Nanger CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From roman at dissauer.net Fri Dec 6 08:30:41 2019 From: roman at dissauer.net (Roman Dissauer) Date: Fri, 6 Dec 2019 09:30:41 +0100 Subject: [Freeswitch-users] t38_gateway hangs up non-deterministic with normal clearing In-Reply-To: References: <255A1D15-BA6C-4D2C-BFE1-BB8B49567ABB@dissauer.net> Message-ID: <7022DD6E-4849-4D67-8984-6EEA2B31E311@dissauer.net> we know that fax over g711 is not the best solution but most of our upstream carriers do not support t.38 but in fact we have approx. 0.7 to 2ms delay over fiber to the upstream crriers TDM switches. We’ve done a lot of testing and we’ve done statistics over several thousand fax attempts with multi page faxes and the result is pretty good at about 95% success rate within the first attempt. Another fact is that if i send a fax through our system as described below and the customer side is another FreeSWITCH it works perfectly! We tested with several t.38 capable pots adapters and it is not really deterministic if the problem exists. First we thought that it only exists if we have several hundred calls on the switch but the issue exists also if there are only 1-3 calls on the switch. Sometimes it works but mostly FreeSWITCH drops the call during the re-Invite. When this issue occurs the call comes via interconnect (A-leg), customer equipment (B-leg) is answering the call. Call is established with g.711a and after about 6 seconds FAX is detected by the end user equipment and an re-Invite is sent back to routing FreeSWITCH. This answers with trying, busy here, bye on the B-leg and also hangs up A-leg with a bye. Screenshot enclosed. I can also pm you the full sip trace of the call. Thanks, Roman > Am 06.12.2019 um 07:56 schrieb Giovanni Maruzzelli : > > Just a couple thoughts : > > - fax over G711 VoIP is totally unreliable, hit or miss, random. It's a fact of nature, no matter how good your internet lines are > - maybe you want try to answer the incoming, wait for 1 second so RTP flow is well established, and then bridge > > -giovanni > > > > On Fri, Dec 6, 2019 at 1:57 AM Roman Dissauer > wrote: > this is our topology: > > customer pbx <-> kamailio + rtpengine <-> several FreeSWITCH Machines <-> kamailio <-> several interconnection carriers > > what we have to address for inbound/outbound calls: > inbound call comes from any interconnection carrier which mostly does NOT have t.38 faxing what is no issue because we have reliable internet lines to our carriers. Our customers want to have t.38 faxing because of reliability of faxing over standard internet lines. So we have to transcode g.711a codec to t.38 in freeswitch. Same is true for outbound calls where customer pbxes send t.38 re-Invite but the upstream interconnection carrier does not accept t.38. > > sometimes faxing is fine with this mechanism but mostly the fax is rejected by a „NORMAL CLEARING“ as soon as the re-Invite is coming in. FreeSWITCH is sending BYE to both call legs. > > what we intend to do with our dialplan on an example - incoming calls > > 1. call is coming in via external profile with g.711a Codec > 2. set refuse_t38=true sets t38 to be rejected coming from a-leg - external carrier > 3. export fax_enable_t38=true will enable t38 both call legs > 4. export nolocal:sip_execute_on_image=t38_image=t38_gateway self nocng will enable t38 only on b-leg - customer side > 5. call is being bridged to kamailio where it gets routed to the customer > > thank you! > > best regards, > Roman > >> Am 04.12.2019 um 14:02 schrieb Giovanni Maruzzelli >: >> >> On Tue, Dec 3, 2019 at 11:19 PM Roman Dissauer > wrote: >> we have an issue with t38_gateway where calls which should be transcoded from PCMA to t38 get hangup non-deterministic. It doesn’t matter if there is only one call or if there are hundrets of calls on the freeswitch. Sometimes it happens that we get several faxes transcoded but then we again have several faxes which don’t go through. problem exists (tested) on FreeSWITCH 1.6 and FreeSWITCH 1.10. >> >> this is the dialplan section where t38_gateway is set up for b-leg on an inbound call: >> >> >> >> >> >> >> >> >> ... >> >> >> >> >> >> >> Would you detail, line by line, what is the intended purpose of each line of dialplan? >> >> Also, your topology, and the intended result >> >> -giovanni >> >> _________________________________________________________________________ >> >> 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 > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > > 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: PastedGraphic-1.png Type: image/png Size: 269331 bytes Desc: not available URL: From m.shirazi at gmail.com Fri Dec 6 06:52:49 2019 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Fri, 6 Dec 2019 10:22:49 +0330 Subject: [Freeswitch-users] Adding user location with ESL Message-ID: Hi Is there a way to add a sip user location info to freeswitch with ESL or fscli (similar to ul_add in opensips) ? Regards M.Shirazi -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Sat Dec 7 00:54:28 2019 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 7 Dec 2019 01:54:28 +0100 Subject: [Freeswitch-users] Long increasing audio delay In-Reply-To: <39aa9728dd444801bf08f084cd24a07f@enghouse.com> References: <39aa9728dd444801bf08f084cd24a07f@enghouse.com> Message-ID: Great! I am greatfull to help community. I am great fan of Windows as I know it best. And also FS likes Windows. I am reading a lot how people have problems with building FS on Linux, but on Windows is matter of seconds. Also there are a lot of good libraries to connect FS ESL with .Net. Anyway, how did I figure it out? Unfortunately there are not a lot of us on Windows to brainstorm about, but leg in audio can be caused only because of timing. And this parameter was invented only because of Linux core. This was my assumption and it worked it out. But I would always suggest using bare metal when go voip and wants to go serious. Best regards, Gregor On Fri, 6 Dec 2019, 20:09 Jaan Kaja, wrote: > Many thanks, Gregor! > > > > It now works like a charm on my bare metal workstation. Audio comes up > fast after call setup, and the audio delay buildup is gone. How did you > figure it out? > > > > Regarding virtualization, a Windows 2016 Server under Hyper-V still has > the problem, but it works fine under VMWare. It may be that people have > done more performance tweaking on the VMWare system. > > > > /Jaan > > > > *From:* FreeSWITCH-users *On > Behalf Of *Gregor Nanger > *Sent:* den 5 december 2019 22:05 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Long increasing audio delay > > > > Hi! > > > > We are using Windows on production for years, also with verto. > > > > As I remember, we had similar problems. I think we solved with timer > settings in verto.conf.xml to: > > > > > > > > Try to set it and see if it works. > > > > Best regards, Gregor > > > > V V čet., 5. dec. 2019 ob 20:10 je oseba Jaan Kaja > napisala: > > Hello, > > > > We are testing webrtc via mod_verto with the verto_communicator demo. > After some initial troubles with certificates, we have it running from > Chrome and Firefox. Very nice! > > > > But, when we leave the connection on for some time, the audio in the > direction from the browser to a phone becomes increasingly longer. We have > clocked ~11 seconds, if we leave it on long enough. The audio in the other > direction has a normal IP telephony delay. We have not found any browser > dependencies. > > > > The FreeSWITCH version is 1.10.1 (latest stable release). The phone end > has been a Blink softphone, or a trunk from a Cisco UCM. The latter > negotiated mulaw PCM with a 20 ms packet size. > > > > We have tried different server OS:es: > > · Windows Server 2019 gives the fastest buildup of delay. > > · Debian installed in a Windows Substrate for Linux gives a > slower buildup of delay, but it does continuously increase. > > · CentOS virtualized in VMWare doesn’t have any buildup of delay, > and the delay is a normal IP telephony delay. > > · So, the more Windows, the worse… > > > > I have a Wireshark trace, and I can get a TLS stream from it. I can > install a key file to decode the stream, but the configuration wants a > protocol, and I don’t know what to enter there. There are no dissectors for > Websocket. So I’m at a bit of a loss, as to how to debug this. I don’t see > anything obvious in the freeswitch.log. > > > > Questions: > > · Has anyone used webrtc on Windows? > > · Has anyone had the same problem? > > · What can be done to debug the problem? Turn on some logging? > > > > We could package a Linux machine to our customers, but as everything else > is on Windows, it would be nice to make it work. > > > > Best regards, > > Jaan > > _________________________________________________________________________ > > 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 > > > > > > -- > > *Gregor Nanger* > > > > *CTO* > t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 > • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia > • www.infomedia.si > > > > _________________________________________________________________________ > > 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 abelcubano at gmail.com Sat Dec 7 22:14:18 2019 From: abelcubano at gmail.com (Abel Monzon) Date: Sat, 7 Dec 2019 17:14:18 -0500 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 Message-ID: Hi. I was trying to get a solution for this but I can't find any online. When FreeSWITCH received an incoming call it send the call trought a provider, but my provider sometimes send OK(200) without 183(Session Progress) and the call is answered. I need from FreeSWITCH to detect this behaviour and send 503 instead of sending OK(200) to the incoming call. This possible? Thank you all for any recommendations Abel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Dec 7 22:58:20 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 7 Dec 2019 22:58:20 +0000 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: I’ve never had to do exactly that, but maybe you can use api_on_media to transfer to another extension where you let it be answered and use bridge_pre_execute_bleg_app to drop the answer and send back a 503 if I’m the call wasn’t transferred. There might be a simpler way, tho. On Sat, 7 Dec 2019 at 22:27, Abel Monzon wrote: > Hi. > > > I was trying to get a solution for this but I can't find any online. > > > When FreeSWITCH received an incoming call it send the call trought a > provider, but my provider sometimes send OK(200) without 183(Session > Progress) and the call is answered. > > I need from FreeSWITCH to detect this behaviour and send 503 instead of > sending OK(200) to the incoming call. > > > This possible? > > Thank you all for any recommendations > > Abel. > _________________________________________________________________________ > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sat Dec 7 23:04:14 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 8 Dec 2019 00:04:14 +0100 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: Why? Provider just answer (200) instead of ringing (18*), why answer with an error (503)? Breaks sip logic... On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: > Hi. > > > I was trying to get a solution for this but I can't find any online. > > > When FreeSWITCH received an incoming call it send the call trought a > provider, but my provider sometimes send OK(200) without 183(Session > Progress) and the call is answered. > > I need from FreeSWITCH to detect this behaviour and send 503 instead of > sending OK(200) to the incoming call. > > > This possible? > > Thank you all for any recommendations > > Abel. > _________________________________________________________________________ > > 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 covici at ccs.covici.com Sun Dec 8 06:03:11 2019 From: covici at ccs.covici.com (John Covici) Date: Sun, 08 Dec 2019 01:03:11 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Message-ID: So, Mike, did you ever figure out what was going on with this? I would love to use the console again for audio? Thanks. On Tue, 03 Dec 2019 18:52:41 -0500, Mike Jerris wrote: > > [1 ] > [1.1 ] > Can you check a pcap to confirm. This MAY be an issue I just saw last week and have to do with rtp timestamps. If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it. > > > On Nov 29, 2019, at 11:28 AM, John Covici wrote: > > > > I have the log of the call which looks normal. My guess is that rtp > > is not properly being sent out, for some reason. The hangup cause is > > always normal_clearing. > > > > On Fri, 29 Nov 2019 13:06:12 -0500, > > David Villasmil wrote: > >> > >> [1 ] > >> [1.1 ] > >> Do you have any trace? > >> > >> On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > >> > >>> Some more information -- even after pressing a digit and getting > >>> audio, it hangs up after about 30 seconds. > >>> > >>> On Fri, 29 Nov 2019 10:46:21 -0500, > >>> John Covici wrote: > >>>> > >>>> Hi. I finally was able to upgrade fs to master as of llast night. > >>>> Its working well, except if I use portaudio to make a call. This all > >>>> worked find in fs 1.6.20. > >>>> > >>>> When I call someone I cannot hear anything until I send it a dtmf > >>>> (rfc2283) and then things work normally, at least I can hear > >>>> something. I had a look at the logs, but nothing strange in there > >>>> after typing the digit. > >>>> > >>>> Also, I cannot call a local extension from port audio, even though the > >>>> extension is registered and can be called from another extension. It > >>>> immediately goes to voicemail. > >>>> > >>>> 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 > >>>> > >>> > >>> -- > >>> 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 > >> > >> -- > >> Regards, > >> > >> David Villasmil > >> email: david.villasmil.work at gmail.com > >> phone: +34669448337 > >> [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 > > > > _________________________________________________________________________ > > > > 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 > [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 david.villasmil.work at gmail.com Sun Dec 8 09:42:04 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 8 Dec 2019 09:42:04 +0000 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: +1 On Sat, 7 Dec 2019 at 23:23, Giovanni Maruzzelli wrote: > Why? > Provider just answer (200) instead of ringing (18*), why answer with an > error (503)? > > Breaks sip logic... > > > > On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: > >> Hi. >> >> >> I was trying to get a solution for this but I can't find any online. >> >> >> When FreeSWITCH received an incoming call it send the call trought a >> provider, but my provider sometimes send OK(200) without 183(Session >> Progress) and the call is answered. >> >> I need from FreeSWITCH to detect this behaviour and send 503 instead of >> sending OK(200) to the incoming call. >> >> >> This possible? >> >> Thank you all for any recommendations >> >> Abel. >> _________________________________________________________________________ >> >> 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From abelcubano at gmail.com Sun Dec 8 17:32:16 2019 From: abelcubano at gmail.com (Abel Monzon) Date: Sun, 8 Dec 2019 12:32:16 -0500 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: FAS is something that in Telecom nobody want, and I detect that if I don't receive ringing (18*) from the provider, them will be a call with FAS and I would like to avoid this for my incoming calls. Make sense? Thank You El sáb., 7 dic. 2019 a las 18:05, Giovanni Maruzzelli () escribió: > Why? > Provider just answer (200) instead of ringing (18*), why answer with an > error (503)? > > Breaks sip logic... > > > > On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: > >> Hi. >> >> >> I was trying to get a solution for this but I can't find any online. >> >> >> When FreeSWITCH received an incoming call it send the call trought a >> provider, but my provider sometimes send OK(200) without 183(Session >> Progress) and the call is answered. >> >> I need from FreeSWITCH to detect this behaviour and send 503 instead of >> sending OK(200) to the incoming call. >> >> >> This possible? >> >> Thank you all for any recommendations >> >> Abel. >> _________________________________________________________________________ >> >> 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 victor.chukalovskiy at gmail.com Sun Dec 8 18:04:24 2019 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Sun, 8 Dec 2019 13:04:24 -0500 Subject: [Freeswitch-users] Git now asking for password Message-ID: Greetings, since a little while git is asking for username / password when I pull before compiling it: > git clone https://freeswitch.org/stash/scm/fs/freeswitch.git > Cloning into 'freeswitch'... > Username for 'https://freeswitch.org': name > Password for 'https://name at freeswitch.org': I have username / password, so not a biggie. But that was not the case before. Is this a new normal now to steer people towards the repo? Or is there a new url to use? Thx! From gmaruzz at gmail.com Sun Dec 8 18:55:12 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 8 Dec 2019 19:55:12 +0100 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: No, at sip level does not make sense. You can implement whitelists, blacklists, call screening ("dial 123 on keypad if you are a human"), etc... That is application logic. I would avoid messing with protocol on a such simple heuristics. But, obviously, free to do it. -giovanni On Sun, Dec 8, 2019, 19:08 Abel Monzon wrote: > FAS is something that in Telecom nobody want, and I detect that if I don't > receive ringing (18*) from the provider, them will be a call with FAS and > I would like to avoid this for my incoming calls. Make sense? > > > Thank You > El sáb., 7 dic. 2019 a las 18:05, Giovanni Maruzzelli () > escribió: > >> Why? >> Provider just answer (200) instead of ringing (18*), why answer with an >> error (503)? >> >> Breaks sip logic... >> >> >> >> On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: >> >>> Hi. >>> >>> >>> I was trying to get a solution for this but I can't find any online. >>> >>> >>> When FreeSWITCH received an incoming call it send the call trought a >>> provider, but my provider sometimes send OK(200) without 183(Session >>> Progress) and the call is answered. >>> >>> I need from FreeSWITCH to detect this behaviour and send 503 instead of >>> sending OK(200) to the incoming call. >>> >>> >>> This possible? >>> >>> Thank you all for any recommendations >>> >>> Abel. >>> _________________________________________________________________________ >>> >>> 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 ciprian.dosoftei at gmail.com Sun Dec 8 19:48:12 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Sun, 8 Dec 2019 14:48:12 -0500 Subject: [Freeswitch-users] Git now asking for password In-Reply-To: References: Message-ID: The project has moved to GitHub a couple of months ago. git clone git at github.com:signalwire/freeswitch.git - or - git clone https://github.com/signalwire/freeswitch.git On Sun, 8 Dec 2019 at 13:25, Victor Chukalovskiy < victor.chukalovskiy at gmail.com> wrote: > Greetings, since a little while git is asking for username / password > when I pull before compiling it: > > > git clone https://freeswitch.org/stash/scm/fs/freeswitch.git > > Cloning into 'freeswitch'... > > Username for 'https://freeswitch.org': name > > Password for 'https://name at freeswitch.org': > > I have username / password, so not a biggie. But that was not the case > before. > > Is this a new normal now to steer people towards the repo? Or is there a > new url to use? > > Thx! > > _________________________________________________________________________ > > 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 -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Dec 8 22:26:52 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 8 Dec 2019 22:26:52 +0000 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: I understand where he's coming from, I understand his conundrum. He doesn't control the termination and it's probably a leaky that doesn't work very well. and there really is no way to filter out FAS calls. If you're positive you've identified calls without 183 will be FAS, i'd say do it. Bear in mind those 200 OK will still be billed to you. But i would do this on Kamailio/Opensips which are better suited for this. With freeswitch it's a duck-tape mess-up. *Of course the right way is to actually have the termination fixed and not having to go around trying to fix by breaking the protocol, but hey.* Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Sun, Dec 8, 2019 at 7:42 PM Giovanni Maruzzelli wrote: > No, at sip level does not make sense. > > You can implement whitelists, blacklists, call screening ("dial 123 on > keypad if you are a human"), etc... > > That is application logic. > > I would avoid messing with protocol on a such simple heuristics. > > But, obviously, free to do it. > > -giovanni > > > > > > On Sun, Dec 8, 2019, 19:08 Abel Monzon wrote: > >> FAS is something that in Telecom nobody want, and I detect that if I >> don't receive ringing (18*) from the provider, them will be a call with >> FAS and I would like to avoid this for my incoming calls. Make sense? >> >> >> Thank You >> El sáb., 7 dic. 2019 a las 18:05, Giovanni Maruzzelli () >> escribió: >> >>> Why? >>> Provider just answer (200) instead of ringing (18*), why answer with an >>> error (503)? >>> >>> Breaks sip logic... >>> >>> >>> >>> On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: >>> >>>> Hi. >>>> >>>> >>>> I was trying to get a solution for this but I can't find any online. >>>> >>>> >>>> When FreeSWITCH received an incoming call it send the call trought a >>>> provider, but my provider sometimes send OK(200) without 183(Session >>>> Progress) and the call is answered. >>>> >>>> I need from FreeSWITCH to detect this behaviour and send 503 instead of >>>> sending OK(200) to the incoming call. >>>> >>>> >>>> This possible? >>>> >>>> Thank you all for any recommendations >>>> >>>> Abel. >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 > > _________________________________________________________________________ > > 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 david.villasmil.work at gmail.com Sun Dec 8 22:46:50 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 8 Dec 2019 22:46:50 +0000 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls Message-ID: Hello all, I'm trying to create a new INVITE using REFER from kamailio. Sending FS the first leg (which is answered in the dialplan and sent to a python script.) And then sending a REFER to FS to call another endpoint. I've tried and i'm always getting *Cannot Blind Transfer 1 Legged calls* and getting a NOTIFY with 404 Not Found Should this work? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From victor.chukalovskiy at gmail.com Sun Dec 8 23:37:25 2019 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Sun, 8 Dec 2019 18:37:25 -0500 Subject: [Freeswitch-users] Git now asking for password In-Reply-To: References: Message-ID: <5cd0765b-e9fa-88eb-cedf-62de80130139@gmail.com> Thank you! On 2019-12-08 2:48 p.m., Ciprian Dosoftei wrote: > The project has moved to GitHub a couple of months ago. > > git clone git at github.com:signalwire/freeswitch.git > > - or - > > git clone https://github.com/signalwire/freeswitch.git > > On Sun, 8 Dec 2019 at 13:25, Victor Chukalovskiy > > > wrote: > > Greetings, since a little while git is asking for username / password > when I pull before compiling it: > > > git clone https://freeswitch.org/stash/scm/fs/freeswitch.git > > Cloning into 'freeswitch'... > > Username for 'https://freeswitch.org': name > > Password for 'https://name at freeswitch.org': > > I have username / password, so not a biggie. But that was not the > case > before. > > Is this a new normal now to steer people towards the repo? Or is > there a > new url to use? > > Thx! > > _________________________________________________________________________ > > 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 > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > > _________________________________________________________________________ > > 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 gmaruzz at gmail.com Mon Dec 9 00:04:16 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 9 Dec 2019 01:04:16 +0100 Subject: [Freeswitch-users] Hangup calls RC=200 without RC=183 In-Reply-To: References: Message-ID: On Mon, Dec 9, 2019, 00:23 David Villasmil wrote: > > > *Of course the right way is to actually have the termination fixed and not > having to go around trying to fix by breaking the protocol, but hey.* > Agreed! > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Sun, Dec 8, 2019 at 7:42 PM Giovanni Maruzzelli > wrote: > >> No, at sip level does not make sense. >> >> You can implement whitelists, blacklists, call screening ("dial 123 on >> keypad if you are a human"), etc... >> >> That is application logic. >> >> I would avoid messing with protocol on a such simple heuristics. >> >> But, obviously, free to do it. >> >> -giovanni >> >> >> >> >> >> On Sun, Dec 8, 2019, 19:08 Abel Monzon wrote: >> >>> FAS is something that in Telecom nobody want, and I detect that if I >>> don't receive ringing (18*) from the provider, them will be a call with >>> FAS and I would like to avoid this for my incoming calls. Make sense? >>> >>> >>> Thank You >>> El sáb., 7 dic. 2019 a las 18:05, Giovanni Maruzzelli (< >>> gmaruzz at gmail.com>) escribió: >>> >>>> Why? >>>> Provider just answer (200) instead of ringing (18*), why answer with an >>>> error (503)? >>>> >>>> Breaks sip logic... >>>> >>>> >>>> >>>> On Sat, Dec 7, 2019, 23:30 Abel Monzon wrote: >>>> >>>>> Hi. >>>>> >>>>> >>>>> I was trying to get a solution for this but I can't find any online. >>>>> >>>>> >>>>> When FreeSWITCH received an incoming call it send the call trought a >>>>> provider, but my provider sometimes send OK(200) without 183(Session >>>>> Progress) and the call is answered. >>>>> >>>>> I need from FreeSWITCH to detect this behaviour and send 503 instead >>>>> of sending OK(200) to the incoming call. >>>>> >>>>> >>>>> This possible? >>>>> >>>>> Thank you all for any recommendations >>>>> >>>>> Abel. >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 >> >> _________________________________________________________________________ >> >> 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 svanherwaarden at precisionag.org Mon Dec 9 09:36:50 2019 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Mon, 9 Dec 2019 10:36:50 +0100 Subject: [Freeswitch-users] SIP handset connection issues Message-ID: Hi, I often have issues where my SIP handset can't seem to connect to my FreeSWITCH server. I'm using Zoiper on iOS and the default freeswitch installation (freeswitch-meta-all) on Debian Buster. The configuration is mostly vanilla - I disabled mod_signalwire, enabled mod_flite and mod_erlang_event, changed the default password and set up an extension that uses mod_erlang_event but that's all. On the server ports 5060, 5070 and 5080 are open on TCP+UDP, ports 3478 and 3479 are open on UDP, and the port range 16384-32768 is also open on UDP. The server runs on a node on DigitalOcean. The issue I'm encountering is that sometimes my handset registers just fine, but other times I get timeout/unreachable errors. Nothing seems to change about the configuration though - I don't restart the server or anything like that. Also, I do see connection attempts on the server from unknown IPs (I'm assuming these are malicious bots trying to get into the system) - this suggests to me that the server is reachable. The system can be unreachable (to me) for days, but then be reachable for days. I have not been able to recognize a pattern. Does anyone have any suggestions what could be going on? Can it be something with my home networking setup? I would be happy to try a different SIP app if anyone has recommendations, it might be easier to debug this problem if I can get more verbose error messages than what Zoiper provides me with. Kind regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Mon Dec 9 19:41:57 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Mon, 9 Dec 2019 14:41:57 -0500 Subject: [Freeswitch-users] SIP handset connection issues In-Reply-To: References: Message-ID: Sam -- I would debug this by validating the premise that your FreeSWITCH node is indeed available through deploying a SIP client on your workstation/laptop using the same user credentials as on your phone. If everything checks out yet Zoiper still fails, it's likely an issue with its networking. For example, perhaps on the days when it works fine the phone might be connected on your local wireless network (and it permits SIP traffic) and when it doesn't, it may be connected through your data plan which might block SIP traffic -- it's a speculation but not an uncommon scenario. If that's the case, the easy fix is to change your SIP port to something nonstandard, and if possible, enable TLS encryption. On Mon, 9 Dec 2019 at 14:32, Sam van Herwaarden < svanherwaarden at precisionag.org> wrote: > Hi, > > I often have issues where my SIP handset can't seem to connect to my > FreeSWITCH server. I'm using Zoiper on iOS and the default freeswitch > installation (freeswitch-meta-all) on Debian Buster. The configuration is > mostly vanilla - I disabled mod_signalwire, enabled mod_flite and > mod_erlang_event, changed the default password and set up an extension that > uses mod_erlang_event but that's all. > > On the server ports 5060, 5070 and 5080 are open on TCP+UDP, ports 3478 > and 3479 are open on UDP, and the port range 16384-32768 is also open on > UDP. The server runs on a node on DigitalOcean. > > The issue I'm encountering is that sometimes my handset registers just > fine, but other times I get timeout/unreachable errors. Nothing seems to > change about the configuration though - I don't restart the server or > anything like that. Also, I do see connection attempts on the server from > unknown IPs (I'm assuming these are malicious bots trying to get into the > system) - this suggests to me that the server is reachable. The system can > be unreachable (to me) for days, but then be reachable for days. I have not > been able to recognize a pattern. > > Does anyone have any suggestions what could be going on? Can it be > something with my home networking setup? I would be happy to try a > different SIP app if anyone has recommendations, it might be easier to > debug this problem if I can get more verbose error messages than what > Zoiper provides me with. > > Kind regards, > Sam > _________________________________________________________________________ > > 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 -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Dec 9 20:16:32 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 9 Dec 2019 20:16:32 +0000 Subject: [Freeswitch-users] SIP handset connection issues In-Reply-To: References: Message-ID: +1 Also, change your Zoiper to TCP instead of UDP. We use thousands of Zoipers and they work well. On Mon, 9 Dec 2019 at 20:03, Ciprian Dosoftei wrote: > Sam -- > > I would debug this by validating the premise that your FreeSWITCH node is > indeed available through deploying a SIP client on your workstation/laptop > using the same user credentials as on your phone. If everything checks out > yet Zoiper still fails, it's likely an issue with its networking. For > example, perhaps on the days when it works fine the phone might be > connected on your local wireless network (and it permits SIP traffic) and > when it doesn't, it may be connected through your data plan which might > block SIP traffic -- it's a speculation but not an uncommon scenario. If > that's the case, the easy fix is to change your SIP port to something > nonstandard, and if possible, enable TLS encryption. > > On Mon, 9 Dec 2019 at 14:32, Sam van Herwaarden < > svanherwaarden at precisionag.org> wrote: > >> Hi, >> >> I often have issues where my SIP handset can't seem to connect to my >> FreeSWITCH server. I'm using Zoiper on iOS and the default freeswitch >> installation (freeswitch-meta-all) on Debian Buster. The configuration is >> mostly vanilla - I disabled mod_signalwire, enabled mod_flite and >> mod_erlang_event, changed the default password and set up an extension that >> uses mod_erlang_event but that's all. >> >> On the server ports 5060, 5070 and 5080 are open on TCP+UDP, ports 3478 >> and 3479 are open on UDP, and the port range 16384-32768 is also open on >> UDP. The server runs on a node on DigitalOcean. >> >> The issue I'm encountering is that sometimes my handset registers just >> fine, but other times I get timeout/unreachable errors. Nothing seems to >> change about the configuration though - I don't restart the server or >> anything like that. Also, I do see connection attempts on the server from >> unknown IPs (I'm assuming these are malicious bots trying to get into the >> system) - this suggests to me that the server is reachable. The system can >> be unreachable (to me) for days, but then be reachable for days. I have not >> been able to recognize a pattern. >> >> Does anyone have any suggestions what could be going on? Can it be >> something with my home networking setup? I would be happy to try a >> different SIP app if anyone has recommendations, it might be easier to >> debug this problem if I can get more verbose error messages than what >> Zoiper provides me with. >> >> Kind regards, >> Sam >> _________________________________________________________________________ >> >> 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 > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Tue Dec 10 13:25:36 2019 From: dragos at freeswitch.org (Dragos Oancea) Date: Tue, 10 Dec 2019 13:25:36 +0000 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Message-ID: Perhaps this solves the problem. https://github.com/signalwire/freeswitch/issues/169 On Sun, Dec 8, 2019 at 6:37 AM John Covici wrote: > So, Mike, did you ever figure out what was going on with this? I > would love to use the console again for audio? > > Thanks. > > On Tue, 03 Dec 2019 18:52:41 -0500, > Mike Jerris wrote: > > > > [1 ] > > [1.1 ] > > Can you check a pcap to confirm. This MAY be an issue I just saw last > week and have to do with rtp timestamps. If you can confirm that rtp is > sending out but the timestamps dont seem right, that would confirm it. > > > > > On Nov 29, 2019, at 11:28 AM, John Covici > wrote: > > > > > > I have the log of the call which looks normal. My guess is that rtp > > > is not properly being sent out, for some reason. The hangup cause is > > > always normal_clearing. > > > > > > On Fri, 29 Nov 2019 13:06:12 -0500, > > > David Villasmil wrote: > > >> > > >> [1 ] > > >> [1.1 ] > > >> Do you have any trace? > > >> > > >> On Fri, 29 Nov 2019 at 18:05, John Covici > wrote: > > >> > > >>> Some more information -- even after pressing a digit and getting > > >>> audio, it hangs up after about 30 seconds. > > >>> > > >>> On Fri, 29 Nov 2019 10:46:21 -0500, > > >>> John Covici wrote: > > >>>> > > >>>> Hi. I finally was able to upgrade fs to master as of llast night. > > >>>> Its working well, except if I use portaudio to make a call. This > all > > >>>> worked find in fs 1.6.20. > > >>>> > > >>>> When I call someone I cannot hear anything until I send it a dtmf > > >>>> (rfc2283) and then things work normally, at least I can hear > > >>>> something. I had a look at the logs, but nothing strange in there > > >>>> after typing the digit. > > >>>> > > >>>> Also, I cannot call a local extension from port audio, even though > the > > >>>> extension is registered and can be called from another extension. > It > > >>>> immediately goes to voicemail. > > >>>> > > >>>> 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 > > >>>> > > >>> > > >>> -- > > >>> 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 > > >> > > >> -- > > >> Regards, > > >> > > >> David Villasmil > > >> email: david.villasmil.work at gmail.com > > >> phone: +34669448337 > > >> [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 > > > > > > > _________________________________________________________________________ > > > > > > 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 FreeSWITCH-users at 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 > > [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 > > _________________________________________________________________________ > > 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 covici at ccs.covici.com Tue Dec 10 14:45:19 2019 From: covici at ccs.covici.com (John Covici) Date: Tue, 10 Dec 2019 09:45:19 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: <369A6EDA-3767-4668-9F7A-231EF87363D3@freeswitch.org> Message-ID: Thanks, I looked on github but saw no actual patch, is that available anywhere? On Tue, 10 Dec 2019 08:25:36 -0500, Dragos Oancea wrote: > > [1 ] > [1.1 ] > Perhaps this solves the problem. > https://github.com/signalwire/freeswitch/issues/169 > > > On Sun, Dec 8, 2019 at 6:37 AM John Covici wrote: > > > So, Mike, did you ever figure out what was going on with this? I > > would love to use the console again for audio? > > > > Thanks. > > > > On Tue, 03 Dec 2019 18:52:41 -0500, > > Mike Jerris wrote: > > > > > > [1 ] > > > [1.1 ] > > > Can you check a pcap to confirm. This MAY be an issue I just saw last > > week and have to do with rtp timestamps. If you can confirm that rtp is > > sending out but the timestamps dont seem right, that would confirm it. > > > > > > > On Nov 29, 2019, at 11:28 AM, John Covici > > wrote: > > > > > > > > I have the log of the call which looks normal. My guess is that rtp > > > > is not properly being sent out, for some reason. The hangup cause is > > > > always normal_clearing. > > > > > > > > On Fri, 29 Nov 2019 13:06:12 -0500, > > > > David Villasmil wrote: > > > >> > > > >> [1 ] > > > >> [1.1 ] > > > >> Do you have any trace? > > > >> > > > >> On Fri, 29 Nov 2019 at 18:05, John Covici > > wrote: > > > >> > > > >>> Some more information -- even after pressing a digit and getting > > > >>> audio, it hangs up after about 30 seconds. > > > >>> > > > >>> On Fri, 29 Nov 2019 10:46:21 -0500, > > > >>> John Covici wrote: > > > >>>> > > > >>>> Hi. I finally was able to upgrade fs to master as of llast night. > > > >>>> Its working well, except if I use portaudio to make a call. This > > all > > > >>>> worked find in fs 1.6.20. > > > >>>> > > > >>>> When I call someone I cannot hear anything until I send it a dtmf > > > >>>> (rfc2283) and then things work normally, at least I can hear > > > >>>> something. I had a look at the logs, but nothing strange in there > > > >>>> after typing the digit. > > > >>>> > > > >>>> Also, I cannot call a local extension from port audio, even though > > the > > > >>>> extension is registered and can be called from another extension. > > It > > > >>>> immediately goes to voicemail. > > > >>>> > > > >>>> 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 > > > >>>> > > > >>> > > > >>> -- > > > >>> 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 > > > >> > > > >> -- > > > >> Regards, > > > >> > > > >> David Villasmil > > > >> email: david.villasmil.work at gmail.com > > > >> phone: +34669448337 > > > >> [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 > > > > > > > > > > _________________________________________________________________________ > > > > > > > > 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 > FreeSWITCH-users at 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 > > > [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 > > > > _________________________________________________________________________ > > > > 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 > [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 jj1330222 at gmail.com Tue Dec 10 16:42:29 2019 From: jj1330222 at gmail.com (Jared J) Date: Tue, 10 Dec 2019 11:42:29 -0500 Subject: [Freeswitch-users] t.38 Fax INCOMPATIBLE_DESTINATION Message-ID: I have inbound faxing working over g.711 however I recently edited the SpanDSP.conf.xml file and added the following lines: Now when I try the same inbound fax that was working before I see this error in the fs_cli [NOTICE] sofia.c:8490 Hangup sofia/internal/5555555555 at example.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] My goal was to have the inbound faxes try t.38 first and then fail back to g.711 ... but obviously that isn't working as expected. Can anyone provide some guidance? -------------- next part -------------- An HTML attachment was scrubbed... URL: From svanherwaarden at precisionag.org Tue Dec 10 09:44:05 2019 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Tue, 10 Dec 2019 10:44:05 +0100 Subject: [Freeswitch-users] SIP handset connection issues In-Reply-To: References: Message-ID: Thanks for the suggestions Ciprian and David. I'm sure that when it wasn't working my phone was connected to WiFi, I experimented with different local networks to see if that made a difference. I will try setting up Zoiper to use TCP and I will experiment with using various clients on my laptop (I have experimented with Linphone a bit but somehow I can't even make it through the account setup assistant - the "Apply" button stays greyed out so I can't register). Best, Sam On Mon, Dec 9, 2019 at 9:17 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > +1 > > Also, change your Zoiper to TCP instead of UDP. > We use thousands of Zoipers and they work well. > > > On Mon, 9 Dec 2019 at 20:03, Ciprian Dosoftei > wrote: > >> Sam -- >> >> I would debug this by validating the premise that your FreeSWITCH node is >> indeed available through deploying a SIP client on your workstation/laptop >> using the same user credentials as on your phone. If everything checks out >> yet Zoiper still fails, it's likely an issue with its networking. For >> example, perhaps on the days when it works fine the phone might be >> connected on your local wireless network (and it permits SIP traffic) and >> when it doesn't, it may be connected through your data plan which might >> block SIP traffic -- it's a speculation but not an uncommon scenario. If >> that's the case, the easy fix is to change your SIP port to something >> nonstandard, and if possible, enable TLS encryption. >> >> On Mon, 9 Dec 2019 at 14:32, Sam van Herwaarden < >> svanherwaarden at precisionag.org> wrote: >> >>> Hi, >>> >>> I often have issues where my SIP handset can't seem to connect to my >>> FreeSWITCH server. I'm using Zoiper on iOS and the default freeswitch >>> installation (freeswitch-meta-all) on Debian Buster. The configuration is >>> mostly vanilla - I disabled mod_signalwire, enabled mod_flite and >>> mod_erlang_event, changed the default password and set up an extension that >>> uses mod_erlang_event but that's all. >>> >>> On the server ports 5060, 5070 and 5080 are open on TCP+UDP, ports 3478 >>> and 3479 are open on UDP, and the port range 16384-32768 is also open on >>> UDP. The server runs on a node on DigitalOcean. >>> >>> The issue I'm encountering is that sometimes my handset registers just >>> fine, but other times I get timeout/unreachable errors. Nothing seems to >>> change about the configuration though - I don't restart the server or >>> anything like that. Also, I do see connection attempts on the server from >>> unknown IPs (I'm assuming these are malicious bots trying to get into the >>> system) - this suggests to me that the server is reachable. The system can >>> be unreachable (to me) for days, but then be reachable for days. I have not >>> been able to recognize a pattern. >>> >>> Does anyone have any suggestions what could be going on? Can it be >>> something with my home networking setup? I would be happy to try a >>> different SIP app if anyone has recommendations, it might be easier to >>> debug this problem if I can get more verbose error messages than what >>> Zoiper provides me with. >>> >>> Kind regards, >>> Sam >>> _________________________________________________________________________ >>> >>> 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 >> >> >> >> -- >> Best Regards, >> Ciprian Dosoftei >> >> The information transmitted is intended only for the addressee and may >> contain privileged and/or confidential material. If you are not the >> intended recipient, kindly contact the sender and delete the message. >> >> Any disclosure, distribution or copying of this message is strictly >> prohibited without the expressed permission of the sender. >> _________________________________________________________________________ >> >> 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 > > -- > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From svanherwaarden at precisionag.org Tue Dec 10 11:13:48 2019 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Tue, 10 Dec 2019 12:13:48 +0100 Subject: [Freeswitch-users] DTMF when I originate a call Message-ID: Hi, I'm building an application that interfaces with FreeSWITCH through mod_erlang_event. For inbound calls (where a user dials my extension) everything works the way I expect. Unfortunately for outbound calls somehow my DTMF events are not coming through. It seems I'm not the first one to experience this issue, but the solutions I find online do not seem to fix the problem for me. This is what my basic originate command looks like (through mod_erlang_event): {foo, freeswitch at fsdev} ! {api, originate, "{origination_caller_id_number=123456,Paddy-Survey=monitoring,Paddy-Question=intro1}user/1003 123456 XML default 1003 1003"}. Based on solutions online I've experimented with adding execute_after_bridge_app=start_dtmf, execute_on_answer=start_dtmf, dtmf_type=info and drop_dtmf=false to the channel variables but none of those seem to help. (The origination caller ID and Paddy-* variables come through as expected so the variables seem to get set.) I've also tried sending the start_dtmf command from my application after answer, I see the command getting executed in the FreeSWITCH log but it doesn't change the behavior (DTMF still doesn't come through). Finally I've tried calling {foo, freeswitch at fsdev} ! {api, uuid_broadcast, CallUUID ++ " start_dtmf"}. after I answer , but this gives me an error 2019-12-10 10:57:18.540690 [ERR] mod_native_file.c:74 Error opening /usr/share/freeswitch/sounds/en/us/callie/start_dtmf.PCMA 2019-12-10 10:57:18.540690 [ERR] mod_sndfile.c:204 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/start_dtmf.wav] [System error : No such file or directory.] Any suggestions on how I can get my outbound calls to work the same way as my inbound calls? Kind regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: From svanherwaarden at precisionag.org Tue Dec 10 14:36:02 2019 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Tue, 10 Dec 2019 15:36:02 +0100 Subject: [Freeswitch-users] DTMF when I originate a call In-Reply-To: References: Message-ID: Actually it turns out that when I use Ekiga on my laptop rather than Zoiper on my iPhone everything just works, without any start_dtmf trickery. Not sure what could be going on there, but it looks like there is nothing wrong with the FreeSWITCH configuration or the way my application controls it. If anyone has suggestions about what could be causing this those would be welcome, but for now I can at least test my controller (that was what I wanted to accomplish). Kind regards, Sam On Tue, Dec 10, 2019 at 12:13 PM Sam van Herwaarden < svanherwaarden at precisionag.org> wrote: > Hi, > > I'm building an application that interfaces with FreeSWITCH through > mod_erlang_event. For inbound calls (where a user dials my extension) > everything works the way I expect. Unfortunately for outbound calls somehow > my DTMF events are not coming through. > > It seems I'm not the first one to experience this issue, but the solutions > I find online do not seem to fix the problem for me. This is what my basic > originate command looks like (through mod_erlang_event): > {foo, freeswitch at fsdev} ! {api, originate, > "{origination_caller_id_number=123456,Paddy-Survey=monitoring,Paddy-Question=intro1}user/1003 > 123456 XML default 1003 1003"}. > > Based on solutions online I've experimented with adding > execute_after_bridge_app=start_dtmf, execute_on_answer=start_dtmf, > dtmf_type=info and drop_dtmf=false to the channel variables but none of > those seem to help. (The origination caller ID and Paddy-* variables come > through as expected so the variables seem to get set.) I've also tried > sending the start_dtmf command from my application after answer, I see > the command getting executed in the FreeSWITCH log but it doesn't change > the behavior (DTMF still doesn't come through). Finally I've tried calling > {foo, freeswitch at fsdev} ! {api, uuid_broadcast, CallUUID ++ " > start_dtmf"}. > after I answer > , > but this gives me an error > 2019-12-10 10:57:18.540690 [ERR] mod_native_file.c:74 Error opening > /usr/share/freeswitch/sounds/en/us/callie/start_dtmf.PCMA > > 2019-12-10 10:57:18.540690 [ERR] mod_sndfile.c:204 Error Opening File > [/usr/share/freeswitch/sounds/en/us/callie/start_dtmf.wav] [System error : > No such file or directory.] > > Any suggestions on how I can get my outbound calls to work the same way as > my inbound calls? > > Kind regards, > Sam > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at freeswitch.com Tue Dec 10 23:05:44 2019 From: alex at freeswitch.com (Alex Sibyakin) Date: Wed, 11 Dec 2019 08:05:44 +0900 Subject: [Freeswitch-users] DTMF when I originate a call In-Reply-To: References: Message-ID: <2f234a96cf4ebce2582d463d993f56545f632f22.camel@freeswitch.com> Hi, Try to enable RFC2833 in Zoiper configuration. https://www.zoiper.com/en/support/answer/for/common/121/DTMF Regards, Alex On Tue, 2019-12-10 at 15:36 +0100, Sam van Herwaarden wrote: > Actually it turns out that when I use Ekiga on my laptop rather than Zoiper on my iPhone everything just works, without any start_dtmf trickery. Not sure what could be going on there, but it looks > like there is nothing wrong with the FreeSWITCH configuration or the way my application controls it. If anyone has suggestions about what could be causing this those would be welcome, but for now I > can at least test my controller (that was what I wanted to accomplish). > > Kind regards, > Sam > > > On Tue, Dec 10, 2019 at 12:13 PM Sam van Herwaarden wrote: > > Hi, > > > > I'm building an application that interfaces with FreeSWITCH through mod_erlang_event. For inbound calls (where a user dials my extension) everything works the way I expect. Unfortunately for > > outbound calls somehow my DTMF events are not coming through. > > > > It seems I'm not the first one to experience this issue, but the solutions I find online do not seem to fix the problem for me. This is what my basic originate command looks like (through > > mod_erlang_event): > > {foo, freeswitch at fsdev} ! {api, originate, "{origination_caller_id_number=123456,Paddy-Survey=monitoring,Paddy-Question=intro1}user/1003 123456 XML default 1003 1003"}. > > > > Based on solutions online I've experimented with adding execute_after_bridge_app=start_dtmf, execute_on_answer=start_dtmf, dtmf_type=info and drop_dtmf=false to the channel variables but none of > > those seem to help. (The origination caller ID and Paddy-* variables come through as expected so the variables seem to get set.) I've also tried sending the start_dtmf command from my application > > after answer, I see the command getting executed in the FreeSWITCH log but it doesn't change the behavior (DTMF still doesn't come through). Finally I've tried calling > > {foo, freeswitch at fsdev} ! {api, uuid_broadcast, CallUUID ++ " start_dtmf"}. > > after I answer, but this gives me an error > > 2019-12-10 10:57:18.540690 [ERR] mod_native_file.c:74 Error opening /usr/share/freeswitch/sounds/en/us/callie/start_dtmf.PCMA > > 2019-12-10 10:57:18.540690 [ERR] mod_sndfile.c:204 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/start_dtmf.wav] [System error : No such file or directory.] > > > > Any suggestions on how I can get my outbound calls to work the same way as my inbound calls? > > > > Kind regards, > > Sam > > _________________________________________________________________________ > > 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 svanherwaarden at precisionag.org Wed Dec 11 08:03:16 2019 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Wed, 11 Dec 2019 09:03:16 +0100 Subject: [Freeswitch-users] DTMF when I originate a call In-Reply-To: <2f234a96cf4ebce2582d463d993f56545f632f22.camel@freeswitch.com> References: <2f234a96cf4ebce2582d463d993f56545f632f22.camel@freeswitch.com> Message-ID: Thanks! It was actually already configured as rfc-2833. But interestingly, switching to "SIP info (numeric)" fixes the issue, now DTMF works both on inbound and outbound calls. Cheers, Sam On Wed, Dec 11, 2019 at 12:06 AM Alex Sibyakin wrote: > Hi, > > Try to enable RFC2833 in Zoiper configuration. > > https://www.zoiper.com/en/support/answer/for/common/121/DTMF > > Regards, > Alex > > > > On Tue, 2019-12-10 at 15:36 +0100, Sam van Herwaarden wrote: > > Actually it turns out that when I use Ekiga on my laptop rather than > Zoiper on my iPhone everything just works, without any start_dtmf trickery. > Not sure what could be going on there, but it looks > > like there is nothing wrong with the FreeSWITCH configuration or the way > my application controls it. If anyone has suggestions about what could be > causing this those would be welcome, but for now I > > can at least test my controller (that was what I wanted to accomplish). > > > > Kind regards, > > Sam > > > > > > On Tue, Dec 10, 2019 at 12:13 PM Sam van Herwaarden < > svanherwaarden at precisionag.org> wrote: > > > Hi, > > > > > > I'm building an application that interfaces with FreeSWITCH through > mod_erlang_event. For inbound calls (where a user dials my extension) > everything works the way I expect. Unfortunately for > > > outbound calls somehow my DTMF events are not coming through. > > > > > > It seems I'm not the first one to experience this issue, but the > solutions I find online do not seem to fix the problem for me. This is what > my basic originate command looks like (through > > > mod_erlang_event): > > > {foo, freeswitch at fsdev} ! {api, originate, > "{origination_caller_id_number=123456,Paddy-Survey=monitoring,Paddy-Question=intro1}user/1003 > 123456 XML default 1003 1003"}. > > > > > > Based on solutions online I've experimented with adding > execute_after_bridge_app=start_dtmf, execute_on_answer=start_dtmf, > dtmf_type=info and drop_dtmf=false to the channel variables but none of > > > those seem to help. (The origination caller ID and Paddy-* variables > come through as expected so the variables seem to get set.) I've also tried > sending the start_dtmf command from my application > > > after answer, I see the command getting executed in the FreeSWITCH log > but it doesn't change the behavior (DTMF still doesn't come through). > Finally I've tried calling > > > {foo, freeswitch at fsdev} ! {api, uuid_broadcast, CallUUID ++ " > start_dtmf"}. > > > after I answer, but this gives me an error > > > 2019-12-10 10:57:18.540690 [ERR] mod_native_file.c:74 Error opening > /usr/share/freeswitch/sounds/en/us/callie/start_dtmf.PCMA > > > > 2019-12-10 10:57:18.540690 [ERR] mod_sndfile.c:204 Error Opening File > [/usr/share/freeswitch/sounds/en/us/callie/start_dtmf.wav] [System error : > No such file or directory.] > > > > > > Any suggestions on how I can get my outbound calls to work the same > way as my inbound calls? > > > > > > Kind regards, > > > Sam > > > > _________________________________________________________________________ > > > > 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 nathan at robotics.net Wed Dec 11 21:04:33 2019 From: nathan at robotics.net (Nathan Stratton) Date: Wed, 11 Dec 2019 16:04:33 -0500 Subject: [Freeswitch-users] Upgrading DTLS Message-ID: Seeing this error on FreeSWITCH 1.10.1 2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is set) but endpoint requires DTLS v1.2. Any idea how to upgrade DTLS to 1.2? I could not find much with a google search. ><> nathan stratton -------------- next part -------------- An HTML attachment was scrubbed... URL: From bruce at bmts.us Wed Dec 11 22:36:02 2019 From: bruce at bmts.us (Bruce Marriner) Date: Wed, 11 Dec 2019 22:36:02 +0000 Subject: [Freeswitch-users] Upgrading from 1.8 to 1.10 on Debian 9 Message-ID: <0101016ef71ad9c4-b1972507-a708-424e-8359-09f3e99681c1-000000@us-west-2.amazonses.com> Hi, I'm sorry if this is documented already, I tried to find my answer on confluence but didn't have any luck. Is it possible and safe to upgrade directly from 1.8 to 1.10 on Debian 9? Is the procedure documented anywhere? Are there any concerns or problems I should expect? (I did see the note about pgsql change). Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Thu Dec 12 08:10:08 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 12 Dec 2019 13:10:08 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Hi, Thanks for your reply, i will check this out, and get back to you Regards Abbasi On Thu, Dec 5, 2019 at 7:38 AM Piotr Gregor wrote: > Module AVMD is optimized for frequencies above 440 Hz. > There is limit hardcoded: > > /*! Minimum beep frequency in Hertz */ > #define AVMD_MIN_FREQUENCY (440.0) > > You can simply change that, and recompile. > BTW: You know voicemails beeping at such low frequencies? 😉 > Wow, it's not beep anymore, it's boop. > > > > Piotr Gregor > Software Engineer > > M: (+44) 07483 866 525 www: dataandsignal.com > > > > > > > On Wed, Dec 4, 2019 at 6:25 PM Bilal Abbasi wrote: > >> Yes Brian, >> My client is looking for something similar to that, but he does not have >> budget to spend around 50$/channel. Dont get me wrong, we love FreeSWITCH, >> but being the dev factory we have to understand client's needs.(and those >> gets very strong, when someone says we need whats freely available in >> Asteriks ;-) ) >> >> On Wed, Dec 4, 2019 at 7:33 PM Brian West wrote: >> >>> You do know we sell a commercial AMD module? >>> >>> On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi >>> wrote: >>> >>>> Hi users, >>>> I am using mod_avmd for quite a while, but i now want to use amd like >>>> its in asterisk. >>>> I found a related module here >>>> https://github.com/seanbright/mod_amd >>>> >>>> I am not able to successfully generate events through that, do someone >>>> has a working dialplan example along with parameters. That would be >>>> really helpful. >>>> >>>> P.S: i have tested on freeswitch 1.10, any special version for this. >>>> >>>> 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 >>> >>> >>> >>> -- >>> >>> 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] >>> _________________________________________________________________________ >>> >>> 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 bilaln018 at gmail.com Thu Dec 12 08:12:03 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 12 Dec 2019 13:12:03 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: But this will still detect beep right? I don't have to detect beep, i want to detect machine (like amd in doing in asterisk). Thats by using some combination of silence and speech signals. On Thu, Dec 5, 2019 at 7:38 AM Piotr Gregor wrote: > Module AVMD is optimized for frequencies above 440 Hz. > There is limit hardcoded: > > /*! Minimum beep frequency in Hertz */ > #define AVMD_MIN_FREQUENCY (440.0) > > You can simply change that, and recompile. > BTW: You know voicemails beeping at such low frequencies? 😉 > Wow, it's not beep anymore, it's boop. > > > > Piotr Gregor > Software Engineer > > M: (+44) 07483 866 525 www: dataandsignal.com > > > > > > > On Wed, Dec 4, 2019 at 6:25 PM Bilal Abbasi wrote: > >> Yes Brian, >> My client is looking for something similar to that, but he does not have >> budget to spend around 50$/channel. Dont get me wrong, we love FreeSWITCH, >> but being the dev factory we have to understand client's needs.(and those >> gets very strong, when someone says we need whats freely available in >> Asteriks ;-) ) >> >> On Wed, Dec 4, 2019 at 7:33 PM Brian West wrote: >> >>> You do know we sell a commercial AMD module? >>> >>> On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi >>> wrote: >>> >>>> Hi users, >>>> I am using mod_avmd for quite a while, but i now want to use amd like >>>> its in asterisk. >>>> I found a related module here >>>> https://github.com/seanbright/mod_amd >>>> >>>> I am not able to successfully generate events through that, do someone >>>> has a working dialplan example along with parameters. That would be >>>> really helpful. >>>> >>>> P.S: i have tested on freeswitch 1.10, any special version for this. >>>> >>>> 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 >>> >>> >>> >>> -- >>> >>> 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] >>> _________________________________________________________________________ >>> >>> 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 mirkobrankovic at gmail.com Thu Dec 12 09:28:33 2019 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Thu, 12 Dec 2019 10:28:33 +0100 Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: I had a same problem, and I see you can set it in vars.conf: https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#L407 but since we have a custom module, it didn't work for me, so I replaced OpenSSL with BorringSSL and fixed it that way :D On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton wrote: > > Seeing this error on FreeSWITCH 1.10.1 > > 2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure > 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is > set) but endpoint requires DTLS v1.2. > > Any idea how to upgrade DTLS to 1.2? I could not find much with a google > search. > > ><> > nathan stratton > _________________________________________________________________________ > > 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 -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From ankit.joban at gmail.com Thu Dec 12 12:25:36 2019 From: ankit.joban at gmail.com (ankit jobanputra) Date: Thu, 12 Dec 2019 17:55:36 +0530 Subject: [Freeswitch-users] No Adminer subcategory found in Default Setting Message-ID: Hi all, I have freshly installed fusionpbx on my Debian 9.5 os. I didn't find the Adminer in Advanced tab, and as the version is 4.2+ I need to do auto-login to true in Default setting - adminer. But I didn't find that setting too. And I tried to add the same setting in Default setting option but I still didn't find the Adminer in advanced tab. Thanks in advance for your help -------------- next part -------------- An HTML attachment was scrubbed... URL: From support at directvoip.co.uk Thu Dec 12 14:19:41 2019 From: support at directvoip.co.uk (Darren Williams) Date: Thu, 12 Dec 2019 15:19:41 +0100 Subject: [Freeswitch-users] No Adminer subcategory found in Default Setting In-Reply-To: References: Message-ID: Adminer has been removed in the latest version but this is a Freeswitch group, you should be asking in FusionPBX IRC or pbxforums.com On Thu, 12 Dec 2019 at 13:54, ankit jobanputra wrote: > Hi all, > > I have freshly installed fusionpbx on my Debian 9.5 os. I didn't find the > Adminer in Advanced tab, and as the version is 4.2+ I need to do auto-login > to true in Default setting - adminer. But I didn't find that setting too. > And I tried to add the same setting in Default setting option but I still > didn't find the Adminer in advanced tab. > > Thanks in advance for your help > > > -- > This message has been scanned for viruses and dangerous content by > *E.F.A. Project* , and is believed to be > clean. > _________________________________________________________________________ > > 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 kaiduanx at yahoo.ca Thu Dec 12 14:47:41 2019 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 12 Dec 2019 14:47:41 +0000 (UTC) Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: <1818241619.7329946.1576162061461@mail.yahoo.com> What OS and version you run FS on? What is the openssl version on the box? /Kaiduan On Thursday, December 12, 2019, 03:29:32 a.m. CST, Mirko Brankovic wrote: I had a same problem, and I see you can set it in vars.conf:https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#L407 but since we have a custom module, it didn't work for me, so I replaced OpenSSL with BorringSSL and fixed it that way :D On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton wrote: Seeing this error on FreeSWITCH 1.10.1  2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is set) but endpoint requires DTLS v1.2. Any idea how to upgrade DTLS to 1.2? I could not find much with a google search. ><> nathan stratton_________________________________________________________________________ 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 -- Regards,Mirko_________________________________________________________________________ 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 mirkobrankovic at gmail.com Thu Dec 12 14:59:41 2019 From: mirkobrankovic at gmail.com (Mirko Brankovic) Date: Thu, 12 Dec 2019 15:59:41 +0100 Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: > > VERSION="16.04.6 LTS (Xenial Xerus)" > ~# dpkg -l | grep openssl > ii libcurl4-openssl-dev:amd64 7.47.0-1ubuntu2.14 > amd64 development files and documentation for > libcurl (OpenSSL flavour) > ii libgnutls-openssl27:amd64 3.4.10-4ubuntu1.5 > amd64 GNU TLS library - OpenSSL wrapper > ii libxmlsec1-openssl 1.2.20-2ubuntu4 > amd64 Openssl engine for the XML security library > ii openssl 1.0.2g-1ubuntu4.15 > amd64 Secure Sockets Layer toolkit - > cryptographic utility But the real problem appeared on another webrtc gateway (Janus) that required TLS 1.2 minimum On Thu, Dec 12, 2019 at 3:48 PM kaiduan xie via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kaiduan xie > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 14:47:41 +0000 (UTC) > Subject: Re: [Freeswitch-users] Upgrading DTLS > What OS and version you run FS on? What is the openssl version on the box? > > /Kaiduan > > On Thursday, December 12, 2019, 03:29:32 a.m. CST, Mirko Brankovic < > mirkobrankovic at gmail.com> wrote: > > > I had a same problem, and I see you can set it in vars.conf: > > https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#L407 > but since we have a custom module, it didn't work for me, so I replaced > OpenSSL with BorringSSL and fixed it that way :D > > On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton > wrote: > > > Seeing this error on FreeSWITCH 1.10.1 > > 2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure > 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is > set) but endpoint requires DTLS v1.2. > > Any idea how to upgrade DTLS to 1.2? I could not find much with a google > search. > > ><> > nathan stratton > _________________________________________________________________________ > > 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 > > > > -- > Regards, > Mirko > _________________________________________________________________________ > > 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 > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 06:48:21 -0800 (PST) > Subject: Re: [Freeswitch-users] Upgrading DTLS > _________________________________________________________________________ > > 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 -- Regards, Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaiduanx at yahoo.ca Thu Dec 12 15:31:40 2019 From: kaiduanx at yahoo.ca (kaiduan xie) Date: Thu, 12 Dec 2019 15:31:40 +0000 (UTC) Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: <105988431.7411034.1576164700485@mail.yahoo.com> Looks like your SSL version is old. #if OPENSSL_VERSION_NUMBER >= 0x10100000 https://github.com/signalwire/freeswitch/blob/master/src/switch_rtp.c#L3757 The following is the SSL version on 16.0.4 Ubuntu. ~:/usr/include/openssl$ grep -R VERSION_NUMBER . -n ./crypto.h:152:# define SSLEAY_VERSION_NUMBER   OPENSSL_VERSION_NUMBER ./ssl.h:2868:# define SSL_R_BAD_PROTOCOL_VERSION_NUMBER                116 ./ssl.h:3164:# define SSL_R_WRONG_VERSION_NUMBER                       267 ./pem.h:589:# define PEM_R_BAD_VERSION_NUMBER                         117 ./opensslv.h:33:# define OPENSSL_VERSION_NUMBER  0x1000207fL ./opensslv.h:83: * The current library version is stored in the macro SHLIB_VERSION_NUMBER, ./opensslv.h:91:# define SHLIB_VERSION_NUMBER "1.0.0" :/usr/include/openssl$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 16.04.5 LTS Release: 16.04 Codename: xenial On Thursday, December 12, 2019, 09:00:42 a.m. CST, Mirko Brankovic wrote: VERSION="16.04.6 LTS (Xenial Xerus)" ~# dpkg -l | grep openssl ii  libcurl4-openssl-dev:amd64       7.47.0-1ubuntu2.14                                        amd64        development files and documentation for libcurl (OpenSSL flavour) ii  libgnutls-openssl27:amd64        3.4.10-4ubuntu1.5                                         amd64        GNU TLS library - OpenSSL wrapper ii  libxmlsec1-openssl               1.2.20-2ubuntu4                                           amd64        Openssl engine for the XML security library ii  openssl                          1.0.2g-1ubuntu4.15                                        amd64        Secure Sockets Layer toolkit - cryptographic utility But the real problem appeared on another webrtc gateway (Janus) that required TLS 1.2 minimum On Thu, Dec 12, 2019 at 3:48 PM kaiduan xie via FreeSWITCH-users wrote: ---------- Forwarded message ---------- From: kaiduan xie To: FreeSWITCH Users Help Cc:  Bcc:  Date: Thu, 12 Dec 2019 14:47:41 +0000 (UTC) Subject: Re: [Freeswitch-users] Upgrading DTLS What OS and version you run FS on? What is the openssl version on the box? /Kaiduan On Thursday, December 12, 2019, 03:29:32 a.m. CST, Mirko Brankovic wrote: I had a same problem, and I see you can set it in vars.conf:https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#L407 but since we have a custom module, it didn't work for me, so I replaced OpenSSL with BorringSSL and fixed it that way :D On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton wrote: Seeing this error on FreeSWITCH 1.10.1  2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is set) but endpoint requires DTLS v1.2. Any idea how to upgrade DTLS to 1.2? I could not find much with a google search. ><> nathan stratton_________________________________________________________________________ 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 -- Regards,Mirko_________________________________________________________________________ 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 ---------- Forwarded message ---------- From: kaiduan xie via FreeSWITCH-users To: FreeSWITCH Users Help Cc:  Bcc:  Date: Thu, 12 Dec 2019 06:48:21 -0800 (PST) Subject: Re: [Freeswitch-users] Upgrading DTLS _________________________________________________________________________ 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 -- Regards,Mirko_________________________________________________________________________ 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 borik.internet at gmail.com Thu Dec 12 16:37:42 2019 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Thu, 12 Dec 2019 19:37:42 +0300 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Hi! You can use freeswitch port of * app_amd, but it still need to some coding... https://github.com/seanbright/mod_amd чт, 12 дек. 2019 г. в 11:53, Bilal Abbasi : > But this will still detect beep right? > I don't have to detect beep, i want to detect machine (like amd in doing > in asterisk). Thats by using some combination of silence and speech signals. > > On Thu, Dec 5, 2019 at 7:38 AM Piotr Gregor > wrote: > >> Module AVMD is optimized for frequencies above 440 Hz. >> There is limit hardcoded: >> >> /*! Minimum beep frequency in Hertz */ >> #define AVMD_MIN_FREQUENCY (440.0) >> >> You can simply change that, and recompile. >> BTW: You know voicemails beeping at such low frequencies? 😉 >> Wow, it's not beep anymore, it's boop. >> >> >> >> Piotr Gregor >> Software Engineer >> >> M: (+44) 07483 866 525 www: dataandsignal.com >> >> >> >> >> >> >> On Wed, Dec 4, 2019 at 6:25 PM Bilal Abbasi wrote: >> >>> Yes Brian, >>> My client is looking for something similar to that, but he does not have >>> budget to spend around 50$/channel. Dont get me wrong, we love FreeSWITCH, >>> but being the dev factory we have to understand client's needs.(and those >>> gets very strong, when someone says we need whats freely available in >>> Asteriks ;-) ) >>> >>> On Wed, Dec 4, 2019 at 7:33 PM Brian West wrote: >>> >>>> You do know we sell a commercial AMD module? >>>> >>>> On Thu, Nov 28, 2019 at 10:09 AM Bilal Abbasi >>>> wrote: >>>> >>>>> Hi users, >>>>> I am using mod_avmd for quite a while, but i now want to use amd like >>>>> its in asterisk. >>>>> I found a related module here >>>>> https://github.com/seanbright/mod_amd >>>>> >>>>> I am not able to successfully generate events through that, do someone >>>>> has a working dialplan example along with parameters. That would be >>>>> really helpful. >>>>> >>>>> P.S: i have tested on freeswitch 1.10, any special version for this. >>>>> >>>>> 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 >>>> >>>> >>>> >>>> -- >>>> >>>> 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] >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 > > _________________________________________________________________________ > > 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 -- With best regards Dmitry Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: From caioebassis at hotmail.com Fri Dec 13 18:23:00 2019 From: caioebassis at hotmail.com (Caio Assis) Date: Fri, 13 Dec 2019 18:23:00 +0000 Subject: [Freeswitch-users] Script timeout Message-ID: Hello everyone. Is it possible to set a timeout of execution of a script? What I mean is if a script takes more than 10 seconds to execute, for example, freeswitch must hangup. I couldn't find anything about it in mod_python documentation. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Fri Dec 13 19:38:57 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Fri, 13 Dec 2019 14:38:57 -0500 Subject: [Freeswitch-users] Script timeout In-Reply-To: References: Message-ID: You could use sched_api (set for your target threshold) to determine whether the conditions for ending the channel are met, and subsequently kill the channel: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-sched_api It really depends on your use case. On Fri, 13 Dec 2019 at 13:57, Caio Assis wrote: > Hello everyone. > > Is it possible to set a timeout of execution of a script? What I mean is > if a script takes more than 10 seconds to execute, for example, freeswitch > must hangup. > > I couldn't find anything about it in mod_python documentation. > > Thanks in advance. > _________________________________________________________________________ > > 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 -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gaurang.gohil at ecosmob.com Mon Dec 16 13:36:50 2019 From: gaurang.gohil at ecosmob.com (Gaurang Gohil) Date: Mon, 16 Dec 2019 19:06:50 +0530 Subject: [Freeswitch-users] No audio with iLBC codec when freeswitch bypass Message-ID: caller codec list : G722,PCMU,iLBC,G729 callee codec list : iLBC,G729 caller------------> FS -----------------> callee iLBC iLBC when callee answered the call iLBC codec negotiated with freeswitch and get proper audio, then after bypass freeswitch using *[uuid_media off ]* no audio coming on both the leg. RTP unassigned after bypass the freeswitch. is there any setting in freeswitch bypass to work with iLBC codec. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalrigel1784 at gmail.com Mon Dec 16 12:16:36 2019 From: vishalrigel1784 at gmail.com (Vishal Dalsania) Date: Mon, 16 Dec 2019 17:46:36 +0530 Subject: [Freeswitch-users] PHP Text to speech Message-ID: Is it possible to implement text to speech via PHP using ESL? I know it ca be done via Lua using mod_lua but not sure if and how it can be done using PHP -------------- next part -------------- An HTML attachment was scrubbed... URL: From bruce at bmts.us Mon Dec 16 15:57:31 2019 From: bruce at bmts.us (Bruce Marriner) Date: Mon, 16 Dec 2019 15:57:31 +0000 Subject: [Freeswitch-users] mod_odbc vs mod_cdr_pg_csv Message-ID: <0101016f0f6dcd65-f75522c7-87ea-454d-90cd-558017e1189d-000000@us-west-2.amazonses.com> Are there any recommendations on which module to use for logging CDR records into PostgreSQL database. It looks like there's two options, mod_odbc and mod_cdr_pg_csv. I'm curious if anyone has had some experience with them and can recommend what they've had the most success with? -------------- next part -------------- An HTML attachment was scrubbed... URL: From jprangi at gmail.com Mon Dec 16 19:28:12 2019 From: jprangi at gmail.com (Jai Rangi) Date: Mon, 16 Dec 2019 11:28:12 -0800 Subject: [Freeswitch-users] G729 License Issue Message-ID: Wondering if anyone else is having the same issue. We got the G729 license reset-ed. Installed the license on new server. (BTW there is new binary for G729 old one does not work and the documentation on https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is outdated) Restarted the license server. But when we try to load the module, it just dont do anything, keep waiting. freeswitch at sip> load mod_com_g729 Keep waiting nothing happens. Press CTRL+c ^CSocket interrupted, bye! Type control-D or /exit or /quit or /bye to exit. Here is my freeswitch version info, status UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, 699 microseconds FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is ready 259 session(s) since startup 9 session(s) - peak 9, last 5min 9 2 session(s) per Sec out of max 500, peak 3, last 5min 2 2000 session(s) max min idle cpu 10.00/97.73 Current Stack Size/Max 240K/8192K What am I missing. If there any change module mod_com_g729, Do I need to recompile/reinstall the module latest source. Have been trying to work with signalwire support with last 10 days but not much help there. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Mon Dec 16 19:56:28 2019 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 16 Dec 2019 12:56:28 -0700 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: References: Message-ID: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> If you dont answer first you can 302 the call, otherwise you can just bridge to the other endpoint. > On Dec 8, 2019, at 3:46 PM, David Villasmil wrote: > > Hello all, > I'm trying to create a new INVITE using REFER from kamailio. > > Sending FS the first leg (which is answered in the dialplan and sent to a python script.) And then sending a REFER to FS to call another endpoint. > > I've tried and i'm always getting Cannot Blind Transfer 1 Legged calls > and getting a NOTIFY with 404 Not Found > > > Should this work? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Dec 16 19:59:38 2019 From: brian at freeswitch.com (Brian West) Date: Mon, 16 Dec 2019 13:59:38 -0600 Subject: [Freeswitch-users] G729 License Issue In-Reply-To: References: Message-ID: You have to install the latest binary from files.freeswitch.org /b On Mon, Dec 16, 2019 at 1:59 PM Jai Rangi wrote: > Wondering if anyone else is having the same issue. > > We got the G729 license reset-ed. Installed the license on new server. > (BTW there is new binary for G729 old one does not work and the > documentation on > https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is > outdated) > > Restarted the license server. But when we try to load the module, it just > dont do anything, keep waiting. > freeswitch at sip> load mod_com_g729 > Keep waiting nothing happens. Press CTRL+c > ^CSocket interrupted, bye! > Type control-D or /exit or /quit or /bye to exit. > > Here is my freeswitch version info, > status > UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, 699 > microseconds > FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is ready > 259 session(s) since startup > 9 session(s) - peak 9, last 5min 9 > 2 session(s) per Sec out of max 500, peak 3, last 5min 2 > 2000 session(s) max > min idle cpu 10.00/97.73 > Current Stack Size/Max 240K/8192K > > What am I missing. If there any change module mod_com_g729, Do I need to > recompile/reinstall the module latest source. > > Have been trying to work with signalwire support with last 10 days but not > much help there. > _________________________________________________________________________ > > 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 david.villasmil.work at gmail.com Mon Dec 16 21:20:09 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 16 Dec 2019 21:20:09 +0000 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: I answer the call with python and keep it there... On Mon, 16 Dec 2019 at 20:51, Mike Jerris wrote: > If you dont answer first you can 302 the call, otherwise you can just > bridge to the other endpoint. > > > On Dec 8, 2019, at 3:46 PM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Hello all, > I'm trying to create a new INVITE using REFER from kamailio. > > Sending FS the first leg (which is answered in the dialplan and sent to a > python script.) And then sending a REFER to FS to call another endpoint. > > I've tried and i'm always getting *Cannot Blind Transfer 1 Legged calls* > and getting a NOTIFY with 404 Not Found > > > Should this work? > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Tue Dec 17 08:55:46 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Tue, 17 Dec 2019 13:55:46 +0500 Subject: [Freeswitch-users] [FreeSWITCH changes Session limits to lower numbers] Message-ID: Hi Users, I am actually doing some stress testing with FreeSWITCH version 1.10. I have set session limit to 1000 and just sending 2CPS but when calls go beyond around 200, it shows me below log and lowers the session limit. Artoo reduces the max sessions to 234 thus, saving the switch from certain doom. 2019-12-16 13:23:07.504847 [CRIT] switch_core_session.c:1818 LUKE'S VOICE: Artoo, see what you can do with it. Hang on back there.... It's 8core 32GB physical machine, and the only change in my FS is that i am using mod_avmd to detect beep. Now what i am really looking for is, if i can disable this behaviour of FreeSWITCH, As it automatically lowers the Session limit to 200-300 range.(which is very low, even if the switch can handle say 500 calls later, it will remain 200-300 sessions). Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From kaduww at gmail.com Tue Dec 17 12:29:27 2019 From: kaduww at gmail.com (Carlos Eduardo) Date: Tue, 17 Dec 2019 09:29:27 -0300 Subject: [Freeswitch-users] [FreeSWITCH changes Session limits to lower numbers] In-Reply-To: References: Message-ID: Hello Bilal, It also happened to me in the past, but I was trying to run more then 4k sessions. It happens because Freeswitch cannot open more threads. I followed the blog belog and it helped me when I had this issue. Basically I set the ulimit -s to 16384 and worked fine. Not it's running 8k sessions with no problem. https://dustycodes.wordpress.com/2012/02/09/increasing-number-of-threads-per-process/ Regards, Em ter., 17 de dez. de 2019 às 05:56, Bilal Abbasi escreveu: > Hi Users, > I am actually doing some stress testing with FreeSWITCH version 1.10. I > have set session limit to 1000 and just sending 2CPS but when calls go > beyond around 200, it shows me below log and lowers the session limit. > > Artoo reduces the max sessions to 234 thus, saving the switch from certain > doom. > 2019-12-16 13:23:07.504847 [CRIT] switch_core_session.c:1818 LUKE'S VOICE: > Artoo, see what you can do with it. Hang on back there.... > > It's 8core 32GB physical machine, and the only change in my FS is that i > am using mod_avmd to detect beep. > Now what i am really looking for is, if i can disable this behaviour of > FreeSWITCH, As it automatically lowers the Session limit to 200-300 > range.(which is very low, even if the switch can handle say 500 calls > later, it will remain 200-300 sessions). > > 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 -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, OCP, dCAA* *E-mail:* *kaduww at gmail.com * *Fone:* +55 48 9981-0894 *Skype:* carlos.e.wagner www.blogdovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalrigel1784 at gmail.com Tue Dec 17 12:56:20 2019 From: vishalrigel1784 at gmail.com (Vishal Dalsania) Date: Tue, 17 Dec 2019 18:26:20 +0530 Subject: [Freeswitch-users] PHP ESL and MySQL database Message-ID: I am able to make PHP ESL work for calls i.e. when i receive a call on specified extension my PHP script does executes and i can control call. However, when i try to access MySQL database from that script, it doesnt let me. Please advise on how i can access database from there. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From abelcubano at gmail.com Wed Dec 18 13:21:07 2019 From: abelcubano at gmail.com (Abel Monzon) Date: Wed, 18 Dec 2019 08:21:07 -0500 Subject: [Freeswitch-users] G729 License Issue In-Reply-To: References: Message-ID: The question is: why you still need mod_com_g729 since codec patent already expired....You should try for free mod_bcg729 Best. El lun., 16 dic. 2019 a las 15:00, Brian West () escribió: > You have to install the latest binary from files.freeswitch.org > > /b > > > On Mon, Dec 16, 2019 at 1:59 PM Jai Rangi wrote: > >> Wondering if anyone else is having the same issue. >> >> We got the G729 license reset-ed. Installed the license on new server. >> (BTW there is new binary for G729 old one does not work and the >> documentation on >> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is >> outdated) >> >> Restarted the license server. But when we try to load the module, it just >> dont do anything, keep waiting. >> freeswitch at sip> load mod_com_g729 >> Keep waiting nothing happens. Press CTRL+c >> ^CSocket interrupted, bye! >> Type control-D or /exit or /quit or /bye to exit. >> >> Here is my freeswitch version info, >> status >> UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, >> 699 microseconds >> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is >> ready >> 259 session(s) since startup >> 9 session(s) - peak 9, last 5min 9 >> 2 session(s) per Sec out of max 500, peak 3, last 5min 2 >> 2000 session(s) max >> min idle cpu 10.00/97.73 >> Current Stack Size/Max 240K/8192K >> >> What am I missing. If there any change module mod_com_g729, Do I need to >> recompile/reinstall the module latest source. >> >> Have been trying to work with signalwire support with last 10 days but >> not much help there. >> _________________________________________________________________________ >> >> 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] > _________________________________________________________________________ > > 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 sagarmalam at gmail.com Wed Dec 18 16:00:01 2019 From: sagarmalam at gmail.com (sagar malam) Date: Wed, 18 Dec 2019 21:30:01 +0530 Subject: [Freeswitch-users] sip_enable_soa variable to false works only when we are using FS in bypass mode ? Message-ID: Hello everyone, Setting sip_enable_soa variable to false works only when we are using FS in bypass mode or media is bypassed after bridge ? I was able to find very limited information about this variable in FS confluence , Any help is appreciated -- Thanks, Sagar -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathan at robotics.net Wed Dec 18 17:37:07 2019 From: nathan at robotics.net (Nathan Stratton) Date: Wed, 18 Dec 2019 12:37:07 -0500 Subject: [Freeswitch-users] Voicemail without video Message-ID: I have set in voicemail.conf.xml but FreeSWITCH is still offering video SDP. We have peer to peer video calls that go to voicemail, but we only want to record audio. ><> nathan stratton -------------- next part -------------- An HTML attachment was scrubbed... URL: From jprangi at gmail.com Wed Dec 18 14:07:29 2019 From: jprangi at gmail.com (Jai Rangi) Date: Wed, 18 Dec 2019 06:07:29 -0800 Subject: [Freeswitch-users] G729 License Issue In-Reply-To: References: Message-ID: For everyone else still using g729 and 1.6 version. It just won’t work even with new binary as well as old. Which I think should work, since g729 is paid for life. I did install from latest, https://files.freeswitch.org/g729/ ls -lrt fs-latest-installer *license* -rwxr-xr-x 1 root freeswitch 4737907 Jun 21 03:25 fs-latest-installer -rwxr-xr-x 1 root freeswitch 3544312 Dec 11 13:22 freeswitch-license-validator root at sip:/usr/local/freeswitch/bin# md5sum fs-latest-installer 9ca7c3cb2f00a8f850605b34e26da402 fs-latest-installer root at sip:/usr/local/freeswitch/bin# md5sum freeswitch-license-validator 113c9be0b6abb8f1f48bb8db6bf0c24a freeswitch-license-validator root at sip:/usr/local/freeswitch/bin/temp#wget https://files.freeswitch.org/g729/fs-latest-installer --2019-12-16 12:12:19-- https://files.freeswitch.org/g729/fs-latest-installer Resolving files.freeswitch.org(files.freeswitch.org)... 190.102.98.174, 2803:d000:fffe::174 Connecting to files.freeswitch.org(files.freeswitch.org)|190.102.98.174|:443... connected. HTTP request sent, awaiting response... 200 OK Length: 4737907 (4.5M) Saving to: ‘fs-latest-installer’ fs-latest-installer 100%[===========================================================================================================================>] 4.52M 5.82MB/s in 0.8s 2019-12-16 12:12:20 (5.82 MB/s) - ‘fs-latest-installer’ saved [4737907 /4737907] root at sip:/usr/local/freeswitch/bin/temp# ls -lrt total 4628 -rw-r--r-- 1 root freeswitch 4737907 Jun 21 03:25 fs-latest-installer root at cbdla5:/usr/local/freeswitch/bin/temp# md5sum fs-latest-installer 9ca7c3cb2f00a8f850605b34e26da402 fs-latest-installer On Wed, Dec 18, 2019 at 5:22 AM Abel Monzon wrote: > The question is: why you still need mod_com_g729 since codec patent > already expired....You should try for free mod_bcg729 > > > Best. > > El lun., 16 dic. 2019 a las 15:00, Brian West () > escribió: > >> You have to install the latest binary from files.freeswitch.org >> >> /b >> >> >> On Mon, Dec 16, 2019 at 1:59 PM Jai Rangi wrote: >> >>> Wondering if anyone else is having the same issue. >>> >>> We got the G729 license reset-ed. Installed the license on new server. >>> (BTW there is new binary for G729 old one does not work and the >>> documentation on >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is >>> outdated) >>> >>> Restarted the license server. But when we try to load the module, it >>> just dont do anything, keep waiting. >>> freeswitch at sip> load mod_com_g729 >>> Keep waiting nothing happens. Press CTRL+c >>> ^CSocket interrupted, bye! >>> Type control-D or /exit or /quit or /bye to exit. >>> >>> Here is my freeswitch version info, >>> status >>> UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, >>> 699 microseconds >>> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is >>> ready >>> 259 session(s) since startup >>> 9 session(s) - peak 9, last 5min 9 >>> 2 session(s) per Sec out of max 500, peak 3, last 5min 2 >>> 2000 session(s) max >>> min idle cpu 10.00/97.73 >>> Current Stack Size/Max 240K/8192K >>> >>> What am I missing. If there any change module mod_com_g729, Do I need to >>> recompile/reinstall the module latest source. >>> >>> Have been trying to work with signalwire support with last 10 days but >>> not much help there. >>> _________________________________________________________________________ >>> >>> 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] >> _________________________________________________________________________ >> >> 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 brian at freeswitch.com Thu Dec 19 12:46:03 2019 From: brian at freeswitch.com (Brian West) Date: Thu, 19 Dec 2019 06:46:03 -0600 Subject: [Freeswitch-users] G729 License Issue In-Reply-To: References: Message-ID: As I stated in my support ticket with you, That 1.6 is no longer supported, We decided to not support it when we rolled out the new binary and keys. 1.6 has been EOL for quite some time. You can install it directly from our repo via 'apt-get install freeswitch-mod-com-g729' on 1.8 or 1.10.x /b On Wed, Dec 18, 2019 at 1:37 PM Jai Rangi wrote: > For everyone else still using g729 and 1.6 version. > It just won’t work even with new binary as well as old. Which I think > should work, since g729 is paid for life. > > I did install from latest, > https://files.freeswitch.org/g729/ > > ls -lrt fs-latest-installer *license* > -rwxr-xr-x 1 root freeswitch 4737907 Jun 21 03:25 fs-latest-installer > -rwxr-xr-x 1 root freeswitch 3544312 Dec 11 13:22 > freeswitch-license-validator > > root at sip:/usr/local/freeswitch/bin# md5sum fs-latest-installer > 9ca7c3cb2f00a8f850605b34e26da402 fs-latest-installer > root at sip:/usr/local/freeswitch/bin# md5sum freeswitch-license-validator > 113c9be0b6abb8f1f48bb8db6bf0c24a freeswitch-license-validator > > > root at sip:/usr/local/freeswitch/bin/temp#wget > https://files.freeswitch.org/g729/fs-latest-installer > --2019-12-16 12:12:19-- > https://files.freeswitch.org/g729/fs-latest-installer > Resolving files.freeswitch.org(files.freeswitch.org)... 190.102.98.174, > 2803:d000:fffe::174 > Connecting to files.freeswitch.org(files.freeswitch.org)|190.102.98.174|:443... > connected. > HTTP request sent, awaiting response... 200 OK > Length: 4737907 (4.5M) > Saving to: ‘fs-latest-installer’ > > fs-latest-installer > 100%[===========================================================================================================================>] > 4.52M 5.82MB/s in 0.8s > > 2019-12-16 12:12:20 (5.82 MB/s) - ‘fs-latest-installer’ saved [4737907 > /4737907] > > root at sip:/usr/local/freeswitch/bin/temp# ls -lrt > total 4628 > -rw-r--r-- 1 root freeswitch 4737907 Jun 21 03:25 fs-latest-installer > root at cbdla5:/usr/local/freeswitch/bin/temp# md5sum fs-latest-installer > 9ca7c3cb2f00a8f850605b34e26da402 fs-latest-installer > > > > > On Wed, Dec 18, 2019 at 5:22 AM Abel Monzon wrote: > >> The question is: why you still need mod_com_g729 since codec patent >> already expired....You should try for free mod_bcg729 >> >> >> Best. >> >> El lun., 16 dic. 2019 a las 15:00, Brian West () >> escribió: >> >>> You have to install the latest binary from files.freeswitch.org >>> >>> /b >>> >>> >>> On Mon, Dec 16, 2019 at 1:59 PM Jai Rangi wrote: >>> >>>> Wondering if anyone else is having the same issue. >>>> >>>> We got the G729 license reset-ed. Installed the license on new server. >>>> (BTW there is new binary for G729 old one does not work and the >>>> documentation on >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is >>>> outdated) >>>> >>>> Restarted the license server. But when we try to load the module, it >>>> just dont do anything, keep waiting. >>>> freeswitch at sip> load mod_com_g729 >>>> Keep waiting nothing happens. Press CTRL+c >>>> ^CSocket interrupted, bye! >>>> Type control-D or /exit or /quit or /bye to exit. >>>> >>>> Here is my freeswitch version info, >>>> status >>>> UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, >>>> 699 microseconds >>>> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is >>>> ready >>>> 259 session(s) since startup >>>> 9 session(s) - peak 9, last 5min 9 >>>> 2 session(s) per Sec out of max 500, peak 3, last 5min 2 >>>> 2000 session(s) max >>>> min idle cpu 10.00/97.73 >>>> Current Stack Size/Max 240K/8192K >>>> >>>> What am I missing. If there any change module mod_com_g729, Do I need >>>> to recompile/reinstall the module latest source. >>>> >>>> Have been trying to work with signalwire support with last 10 days but >>>> not much help there. >>>> >>>> _________________________________________________________________________ >>>> >>>> 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] >>> _________________________________________________________________________ >>> >>> 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 -- 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 brian at freeswitch.com Thu Dec 19 12:46:37 2019 From: brian at freeswitch.com (Brian West) Date: Thu, 19 Dec 2019 06:46:37 -0600 Subject: [Freeswitch-users] G729 License Issue In-Reply-To: References: Message-ID: Don't confuse copyright and patent, just because the patent has expired, doesn't mean the copyright has expired on the code. /b On Wed, Dec 18, 2019 at 7:41 AM Abel Monzon wrote: > The question is: why you still need mod_com_g729 since codec patent > already expired....You should try for free mod_bcg729 > > > Best. > > El lun., 16 dic. 2019 a las 15:00, Brian West () > escribió: > >> You have to install the latest binary from files.freeswitch.org >> >> /b >> >> >> On Mon, Dec 16, 2019 at 1:59 PM Jai Rangi wrote: >> >>> Wondering if anyone else is having the same issue. >>> >>> We got the G729 license reset-ed. Installed the license on new server. >>> (BTW there is new binary for G729 old one does not work and the >>> documentation on >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729 is >>> outdated) >>> >>> Restarted the license server. But when we try to load the module, it >>> just dont do anything, keep waiting. >>> freeswitch at sip> load mod_com_g729 >>> Keep waiting nothing happens. Press CTRL+c >>> ^CSocket interrupted, bye! >>> Type control-D or /exit or /quit or /bye to exit. >>> >>> Here is my freeswitch version info, >>> status >>> UP 0 years, 0 days, 0 hours, 10 minutes, 17 seconds, 610 milliseconds, >>> 699 microseconds >>> FreeSWITCH (Version 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit) is >>> ready >>> 259 session(s) since startup >>> 9 session(s) - peak 9, last 5min 9 >>> 2 session(s) per Sec out of max 500, peak 3, last 5min 2 >>> 2000 session(s) max >>> min idle cpu 10.00/97.73 >>> Current Stack Size/Max 240K/8192K >>> >>> What am I missing. If there any change module mod_com_g729, Do I need to >>> recompile/reinstall the module latest source. >>> >>> Have been trying to work with signalwire support with last 10 days but >>> not much help there. >>> _________________________________________________________________________ >>> >>> 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] >> _________________________________________________________________________ >> >> 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 -- 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 yehavi.bourvine at gmail.com Sun Dec 22 05:35:16 2019 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 22 Dec 2019 07:35:16 +0200 Subject: [Freeswitch-users] SIP trunk Cisco-Freeswitch: Cannot return from HOLD Message-ID: Hello, I am connecting FS 1.8 with Cisco CUCM. A call is initiated correctly, and there is two way audio. When the Cisco phone places the call on hold, cisco sends SDP with "sendonly". Upon return from HOLD, Cisco sends INVITE without SDP and FS replies with SDP with "sendonly"... As a result, there is a one way audio from FS to Cisco. I can roll this problem to Cisco as it does not cancels the "sendonly" from its side, but why FS sends also "sendonly"? This probably causes Cisco to not send audio anymore. Anyone noticed this problem and has a solution? Thanks, __yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sun Dec 22 06:40:12 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 22 Dec 2019 09:40:12 +0300 Subject: [Freeswitch-users] SIP trunk Cisco-Freeswitch: Cannot return from HOLD In-Reply-To: References: Message-ID: Think if devices send INVITE without SDP, then they must send SDP in ACK message. Please check. On Sun, Dec 22, 2019 at 9:02 AM Yehavi Bourvine wrote: > Hello, > I am connecting FS 1.8 with Cisco CUCM. A call is initiated correctly, > and there is two way audio. When the Cisco phone places the call on hold, > cisco sends SDP with "sendonly". Upon return from HOLD, Cisco sends INVITE > without SDP and FS replies with SDP with "sendonly"... As a result, there > is a one way audio from FS to Cisco. I can roll this problem to Cisco as it > does not cancels the "sendonly" from its side, but why FS sends also > "sendonly"? This probably causes Cisco to not send audio anymore. > > Anyone noticed this problem and has a solution? > > Thanks, __yehavi: > _________________________________________________________________________ > > 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 yehavi.bourvine at gmail.com Sun Dec 22 08:04:24 2019 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 22 Dec 2019 10:04:24 +0200 Subject: [Freeswitch-users] SIP trunk Cisco-Freeswitch: Cannot return from HOLD In-Reply-To: References: Message-ID: Hi, Cisco sends the SDP in the ACK. The problem is that Cisco sends 'sendonly" in SDP and FS replies also with "sendonly", and there is no apparent reason for this. This causes then a one-way audio. Thanks, __Yehavi: ‫בתאריך יום א׳, 22 בדצמ׳ 2019 ב-8:41 מאת ‪Sergey Safarov‬‏ <‪ s.safarov at gmail.com‬‏>:‬ > Think if devices send INVITE without SDP, then they must send SDP in ACK > message. > Please check. > > On Sun, Dec 22, 2019 at 9:02 AM Yehavi Bourvine > wrote: > >> Hello, >> I am connecting FS 1.8 with Cisco CUCM. A call is initiated correctly, >> and there is two way audio. When the Cisco phone places the call on hold, >> cisco sends SDP with "sendonly". Upon return from HOLD, Cisco sends INVITE >> without SDP and FS replies with SDP with "sendonly"... As a result, there >> is a one way audio from FS to Cisco. I can roll this problem to Cisco as it >> does not cancels the "sendonly" from its side, but why FS sends also >> "sendonly"? This probably causes Cisco to not send audio anymore. >> >> Anyone noticed this problem and has a solution? >> >> Thanks, __yehavi: >> _________________________________________________________________________ >> >> 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 s.safarov at gmail.com Sun Dec 22 08:45:24 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Sun, 22 Dec 2019 11:45:24 +0300 Subject: [Freeswitch-users] SIP trunk Cisco-Freeswitch: Cannot return from HOLD In-Reply-To: References: Message-ID: yes looks as bug, please report https://github.com/signalwire/freeswitch/issues https://tools.ietf.org/html/rfc6337#section-5.3 5.3 . Hold and Resume of Media [RFC3264] specifies (using non-normative language) that "hold" should be indicated in an established session by sending a new offer containing "a=sendonly" attribute for each media stream to be held. An answerer is then to respond with "a=recvonly" attribute to acknowledge that the hold request has been understood. On Sun, Dec 22, 2019 at 11:27 AM Yehavi Bourvine wrote: > Hi, > > Cisco sends the SDP in the ACK. The problem is that Cisco sends > 'sendonly" in SDP and FS replies also with "sendonly", and there is no > apparent reason for this. This causes then a one-way audio. > > Thanks, __Yehavi: > > ‫בתאריך יום א׳, 22 בדצמ׳ 2019 ב-8:41 מאת ‪Sergey Safarov‬‏ <‪ > s.safarov at gmail.com‬‏>:‬ > >> Think if devices send INVITE without SDP, then they must send SDP in ACK >> message. >> Please check. >> >> On Sun, Dec 22, 2019 at 9:02 AM Yehavi Bourvine < >> yehavi.bourvine at gmail.com> wrote: >> >>> Hello, >>> I am connecting FS 1.8 with Cisco CUCM. A call is initiated correctly, >>> and there is two way audio. When the Cisco phone places the call on hold, >>> cisco sends SDP with "sendonly". Upon return from HOLD, Cisco sends INVITE >>> without SDP and FS replies with SDP with "sendonly"... As a result, there >>> is a one way audio from FS to Cisco. I can roll this problem to Cisco as it >>> does not cancels the "sendonly" from its side, but why FS sends also >>> "sendonly"? This probably causes Cisco to not send audio anymore. >>> >>> Anyone noticed this problem and has a solution? >>> >>> Thanks, __yehavi: >>> _________________________________________________________________________ >>> >>> 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 borik.internet at gmail.com Tue Dec 24 21:09:26 2019 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Wed, 25 Dec 2019 00:09:26 +0300 Subject: [Freeswitch-users] detect_speech unimrcp ... ends with 'Aborted (core dumped)' Message-ID: Hi, All! I have a strange problem... I have a big Lua script to initiate a call. It starts detect_speech with > execute_on_media=detect_speech unimrcp ... Script initiate Leg B on leg A answer. And when leg B answers FreeSWITCH halted with this message > 2019-12-25 00:00:55.903734 [INFO] switch_ivr_async.c:219 Digit parser > DPTOOLS: Setting realm to 'temp' > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_async.c:344 Digit parser > DPTOOLS: binding *#/temp/0 callback: 0x7fb375cf9700 data: 0x7fb2fc119818 > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1672 > (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State Change > CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:584 > (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) Running State Change > CS_CONSUME_MEDIA (Cur 2 Tot 2) > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA > 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1063 sofia/internal/ > 1-FROM_GEN at 10.17.19.111:5060 CUSTOM HOLD > 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA going > to sleep > freeswitch: src/switch_buffer.c:348: switch_buffer_zero: Assertion > `buffer->data != ((void *)0)' failed. > Aborted (core dumped) > I need to fix it ASAP, but I don't know where to see at all! What is it can be? -- With best regards Dmitry Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Wed Dec 25 00:26:13 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 25 Dec 2019 03:26:13 +0300 Subject: [Freeswitch-users] detect_speech unimrcp ... ends with 'Aborted (core dumped)' In-Reply-To: References: Message-ID: Looks as buffer not initialized. Need investigate backtrace. Please create ticket here https://github.com/signalwire/freeswitch/issues On Wed, Dec 25, 2019 at 12:39 AM Dmitriy Borisov wrote: > Hi, All! > > I have a strange problem... I have a big Lua script to initiate a call. It > starts detect_speech with > >> execute_on_media=detect_speech unimrcp ... > > Script initiate Leg B on leg A answer. And when leg B answers FreeSWITCH > halted with this message > >> 2019-12-25 00:00:55.903734 [INFO] switch_ivr_async.c:219 Digit parser >> DPTOOLS: Setting realm to 'temp' >> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_async.c:344 Digit parser >> DPTOOLS: binding *#/temp/0 callback: 0x7fb375cf9700 data: 0x7fb2fc119818 >> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1672 >> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State Change >> CS_SOFT_EXECUTE -> CS_CONSUME_MEDIA >> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:584 >> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) Running State Change >> CS_CONSUME_MEDIA (Cur 2 Tot 2) >> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA >> 2019-12-25 00:00:55.903734 [DEBUG] switch_ivr_bridge.c:1063 >> sofia/internal/1-FROM_GEN at 10.17.19.111:5060 CUSTOM HOLD >> 2019-12-25 00:00:55.903734 [DEBUG] switch_core_state_machine.c:662 >> (sofia/internal/1-FROM_GEN at 10.17.19.111:5060) State CONSUME_MEDIA going >> to sleep >> freeswitch: src/switch_buffer.c:348: switch_buffer_zero: Assertion >> `buffer->data != ((void *)0)' failed. >> Aborted (core dumped) >> > I need to fix it ASAP, but I don't know where to see at all! What is it > can be? > > -- > With best regards > Dmitry Borisov > _________________________________________________________________________ > > 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 yehavi.bourvine at gmail.com Wed Dec 25 07:27:58 2019 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 25 Dec 2019 09:27:58 +0200 Subject: [Freeswitch-users] SIP trunk Cisco-Freeswitch: Cannot return from HOLD In-Reply-To: References: Message-ID: It seems that the problem is already known and FS developers gave a workaround: https://freeswitch.org/jira/browse/FS-9765 Actually, Cisco is working incorrectly, but FreeSwitch is more flexible... Thanks, __Yehavi: ‫בתאריך יום א׳, 22 בדצמ׳ 2019 ב-10:46 מאת ‪Sergey Safarov‬‏ <‪ s.safarov at gmail.com‬‏>:‬ > yes looks as bug, please report > https://github.com/signalwire/freeswitch/issues > > https://tools.ietf.org/html/rfc6337#section-5.3 > > 5.3 . Hold and Resume of Media > > [RFC3264] specifies (using non-normative language) that "hold" should > be indicated in an established session by sending a new offer > containing "a=sendonly" attribute for each media stream to be held. > An answerer is then to respond with "a=recvonly" attribute to > acknowledge that the hold request has been understood. > > > > > On Sun, Dec 22, 2019 at 11:27 AM Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> Hi, >> >> Cisco sends the SDP in the ACK. The problem is that Cisco sends >> 'sendonly" in SDP and FS replies also with "sendonly", and there is no >> apparent reason for this. This causes then a one-way audio. >> >> Thanks, __Yehavi: >> >> ‫בתאריך יום א׳, 22 בדצמ׳ 2019 ב-8:41 מאת ‪Sergey Safarov‬‏ <‪ >> s.safarov at gmail.com‬‏>:‬ >> >>> Think if devices send INVITE without SDP, then they must send SDP in ACK >>> message. >>> Please check. >>> >>> On Sun, Dec 22, 2019 at 9:02 AM Yehavi Bourvine < >>> yehavi.bourvine at gmail.com> wrote: >>> >>>> Hello, >>>> I am connecting FS 1.8 with Cisco CUCM. A call is initiated >>>> correctly, and there is two way audio. When the Cisco phone places the call >>>> on hold, cisco sends SDP with "sendonly". Upon return from HOLD, Cisco >>>> sends INVITE without SDP and FS replies with SDP with "sendonly"... As a >>>> result, there is a one way audio from FS to Cisco. I can roll this problem >>>> to Cisco as it does not cancels the "sendonly" from its side, but why FS >>>> sends also "sendonly"? This probably causes Cisco to not send audio anymore. >>>> >>>> Anyone noticed this problem and has a solution? >>>> >>>> Thanks, __yehavi: >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 > > _________________________________________________________________________ > > 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 vishalmpai at gmail.com Thu Dec 26 18:04:56 2019 From: vishalmpai at gmail.com (Vishal Pai) Date: Thu, 26 Dec 2019 23:34:56 +0530 Subject: [Freeswitch-users] PHP ESL Installation Message-ID: Hello everyone Season's Greetings. I am having php7.0 installed on my Debian 9 OS Using below command apt install php7.0 libapache2-mod-php7.0 php7.0-common php7.0-pdo php7.0-mbstring php7.0-xmlrpc php7.0-soap php7.0-gd php7.0-xml php7.0-intl php7.0-mysql php7.0-cli php7.0-mcrypt php7.0-ldap php7.0-zip php7.0-curl php7.0-dev Now when i compile esl under /usr/src/freeswitch-1.10.1.-release/libs/esl make phpmod-install i am getting following errors make MYLIB=".././.libs/libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/include/uuid -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -Werror -Wno-unused-result -Wno-misleading-indentation -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL" CXXFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1" CXX_CFLAGS="" -C php make[1]: Entering directory '/usr/src/freeswitch-1.10.1.-release/libs/esl/php' g++ -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/src/include -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/include/php/20151012 -I/usr/include/php/20151012/main -I/usr/include/php/20151012/TSRM -I/usr/include/php/20151012/Zend -I/usr/include/php/20151012/ext -I/usr/include/php/20151012/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp:876:105: error: macro "zend_hash_update" passed 6 arguments, but takes just 3 zend_hash_update(HASH_OF(z), (char*)"_cPtr", sizeof("_cPtr"), (void*)&resource, sizeof(zval), NULL); esl_wrap.cpp:1246:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1521:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1610:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1640:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1670:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1918:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp:1948:50: error: macro "ZVAL_STRING" passed 3 arguments, but takes just 2 ZVAL_STRING(return_value, (char *)result, 1); ^ esl_wrap.cpp: In function ‘void SWIG_landfill(zend_resource*)’: esl_wrap.cpp:813:51: error: ‘rsrc’ was not declared in this scope static ZEND_RSRC_DTOR_FUNC(SWIG_landfill) { (void)rsrc; } ^~~~ esl_wrap.cpp: In function ‘void SWIG_ZTS_SetPointerZval(zval*, void*, swig_type_info*, int)’: esl_wrap.cpp:836:66: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in this scope ZEND_REGISTER_RESOURCE(z, value, *(int *)(type->clientdata)); ^ esl_wrap.cpp:857:29: error: ‘MAKE_STD_ZVAL’ was not declared in this scope MAKE_STD_ZVAL(resource); ^ esl_wrap.cpp:858:73: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in this scope ZEND_REGISTER_RESOURCE(resource, value, *(int *)(type->clientdata)); ^ esl_wrap.cpp:863:93: error: cannot convert ‘char*’ to ‘zend_string* {aka _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* zend_lookup_class(zend_string*)’ result = zend_lookup_class(classname, SWIG_PREFIX_LEN + type_name_len, &ce TSRMLS_CC); ^ esl_wrap.cpp:866:83: error: cannot convert ‘char*’ to ‘zend_string* {aka _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* zend_lookup_class(zend_string*)’ result = zend_lookup_class((char *)type_name, type_name_len, &ce TSRMLS_CC); Which doesn't get compiled successfully. Can anyone let me know what i am doing wrong here. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Thu Dec 26 18:40:29 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Thu, 26 Dec 2019 13:40:29 -0500 Subject: [Freeswitch-users] PHP ESL Installation In-Reply-To: References: Message-ID: You're using the PHP 7+ development packages against an extension designed for PHP 5 (deprecated). This has been addressed after 1.10.1 was released so you can try building the PHP extension against the latest master or any commit after 93691c60cea1c2e873b0507134815500401ca9cb. On Thu, 26 Dec 2019 at 13:24, Vishal Pai wrote: > Hello everyone > > Season's Greetings. > > I am having php7.0 installed on my Debian 9 OS Using below command > > apt install php7.0 libapache2-mod-php7.0 php7.0-common php7.0-pdo > php7.0-mbstring php7.0-xmlrpc php7.0-soap php7.0-gd php7.0-xml php7.0-intl > php7.0-mysql php7.0-cli php7.0-mcrypt php7.0-ldap php7.0-zip php7.0-curl > php7.0-dev > > Now when i compile esl under > > /usr/src/freeswitch-1.10.1.-release/libs/esl make phpmod-install > > i am getting following errors > > make MYLIB=".././.libs/libesl.a" SOLINK="-shared -Xlinker -x" > CFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include > -I/usr/include/uuid -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC > -ffast-math -Werror -Wno-unused-result -Wno-misleading-indentation > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL" > CXXFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC > -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1" CXX_CFLAGS="" -C php > make[1]: Entering directory > '/usr/src/freeswitch-1.10.1.-release/libs/esl/php' > g++ -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC > -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/src/include > -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC > -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/include/php/20151012 > -I/usr/include/php/20151012/main -I/usr/include/php/20151012/TSRM > -I/usr/include/php/20151012/Zend -I/usr/include/php/20151012/ext > -I/usr/include/php/20151012/ext/date/lib -Wno-unused-label > -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp:876:105: error: macro "zend_hash_update" passed 6 arguments, > but takes just 3 > zend_hash_update(HASH_OF(z), (char*)"_cPtr", sizeof("_cPtr"), > (void*)&resource, sizeof(zval), NULL); > esl_wrap.cpp:1246:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1521:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1610:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1640:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1670:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1918:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp:1948:50: error: macro "ZVAL_STRING" passed 3 arguments, but > takes just 2 > ZVAL_STRING(return_value, (char *)result, 1); > ^ > esl_wrap.cpp: In function ‘void SWIG_landfill(zend_resource*)’: > esl_wrap.cpp:813:51: error: ‘rsrc’ was not declared in this scope > static ZEND_RSRC_DTOR_FUNC(SWIG_landfill) { (void)rsrc; } > ^~~~ > esl_wrap.cpp: In function ‘void SWIG_ZTS_SetPointerZval(zval*, void*, > swig_type_info*, int)’: > esl_wrap.cpp:836:66: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in > this scope > ZEND_REGISTER_RESOURCE(z, value, *(int *)(type->clientdata)); > ^ > esl_wrap.cpp:857:29: error: ‘MAKE_STD_ZVAL’ was not declared in this scope > MAKE_STD_ZVAL(resource); > ^ > esl_wrap.cpp:858:73: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in > this scope > ZEND_REGISTER_RESOURCE(resource, value, *(int *)(type->clientdata)); > ^ > esl_wrap.cpp:863:93: error: cannot convert ‘char*’ to ‘zend_string* {aka > _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* > zend_lookup_class(zend_string*)’ > result = zend_lookup_class(classname, SWIG_PREFIX_LEN + > type_name_len, &ce TSRMLS_CC); > > ^ > esl_wrap.cpp:866:83: error: cannot convert ‘char*’ to ‘zend_string* {aka > _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* > zend_lookup_class(zend_string*)’ > result = zend_lookup_class((char *)type_name, type_name_len, &ce > TSRMLS_CC); > > > Which doesn't get compiled successfully. Can anyone let me know what i am > doing wrong here. > > _________________________________________________________________________ > > 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 -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Fri Dec 27 13:30:50 2019 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 27 Dec 2019 19:00:50 +0530 Subject: [Freeswitch-users] PHP ESL Installation In-Reply-To: References: Message-ID: Thank you Ciprian Dosoftei. I was able to install with version latest development version FreeSWITCH (Version 1.10.2-dev 64bit). On Fri, Dec 27, 2019 at 12:30 AM Ciprian Dosoftei < ciprian.dosoftei at gmail.com> wrote: > You're using the PHP 7+ development packages against an extension designed > for PHP 5 (deprecated). > > This has been addressed after 1.10.1 was released so you can try building > the PHP extension against the latest master or any commit > after 93691c60cea1c2e873b0507134815500401ca9cb. > > On Thu, 26 Dec 2019 at 13:24, Vishal Pai wrote: > >> Hello everyone >> >> Season's Greetings. >> >> I am having php7.0 installed on my Debian 9 OS Using below command >> >> apt install php7.0 libapache2-mod-php7.0 php7.0-common php7.0-pdo >> php7.0-mbstring php7.0-xmlrpc php7.0-soap php7.0-gd php7.0-xml php7.0-intl >> php7.0-mysql php7.0-cli php7.0-mcrypt php7.0-ldap php7.0-zip php7.0-curl >> php7.0-dev >> >> Now when i compile esl under >> >> /usr/src/freeswitch-1.10.1.-release/libs/esl make phpmod-install >> >> i am getting following errors >> >> make MYLIB=".././.libs/libesl.a" SOLINK="-shared -Xlinker -x" >> CFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include >> -I/usr/include/uuid -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC >> -ffast-math -Werror -Wno-unused-result -Wno-misleading-indentation >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DCJSON_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL" >> CXXFLAGS="-I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC >> -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1" CXX_CFLAGS="" -C php >> make[1]: Entering directory >> '/usr/src/freeswitch-1.10.1.-release/libs/esl/php' >> g++ -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC >> -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >> -I/usr/src/freeswitch-1.10.1.-release/libs/esl/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/src/include >> -I/usr/src/freeswitch-1.10.1.-release/libs/libteletone/src -fPIC >> -ffast-math -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DCJSON_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/include/php/20151012 >> -I/usr/include/php/20151012/main -I/usr/include/php/20151012/TSRM >> -I/usr/include/php/20151012/Zend -I/usr/include/php/20151012/ext >> -I/usr/include/php/20151012/ext/date/lib -Wno-unused-label >> -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o >> esl_wrap.cpp:876:105: error: macro "zend_hash_update" passed 6 arguments, >> but takes just 3 >> zend_hash_update(HASH_OF(z), (char*)"_cPtr", sizeof("_cPtr"), >> (void*)&resource, sizeof(zval), NULL); >> esl_wrap.cpp:1246:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1521:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1610:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1640:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1670:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1918:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp:1948:50: error: macro "ZVAL_STRING" passed 3 arguments, but >> takes just 2 >> ZVAL_STRING(return_value, (char *)result, 1); >> ^ >> esl_wrap.cpp: In function ‘void SWIG_landfill(zend_resource*)’: >> esl_wrap.cpp:813:51: error: ‘rsrc’ was not declared in this scope >> static ZEND_RSRC_DTOR_FUNC(SWIG_landfill) { (void)rsrc; } >> ^~~~ >> esl_wrap.cpp: In function ‘void SWIG_ZTS_SetPointerZval(zval*, void*, >> swig_type_info*, int)’: >> esl_wrap.cpp:836:66: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in >> this scope >> ZEND_REGISTER_RESOURCE(z, value, *(int *)(type->clientdata)); >> ^ >> esl_wrap.cpp:857:29: error: ‘MAKE_STD_ZVAL’ was not declared in this scope >> MAKE_STD_ZVAL(resource); >> ^ >> esl_wrap.cpp:858:73: error: ‘ZEND_REGISTER_RESOURCE’ was not declared in >> this scope >> ZEND_REGISTER_RESOURCE(resource, value, *(int >> *)(type->clientdata)); >> ^ >> esl_wrap.cpp:863:93: error: cannot convert ‘char*’ to ‘zend_string* {aka >> _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* >> zend_lookup_class(zend_string*)’ >> result = zend_lookup_class(classname, SWIG_PREFIX_LEN + >> type_name_len, &ce TSRMLS_CC); >> >> ^ >> esl_wrap.cpp:866:83: error: cannot convert ‘char*’ to ‘zend_string* {aka >> _zend_string*}’ for argument ‘1’ to ‘zend_class_entry* >> zend_lookup_class(zend_string*)’ >> result = zend_lookup_class((char *)type_name, type_name_len, &ce >> TSRMLS_CC); >> >> >> Which doesn't get compiled successfully. Can anyone let me know what i am >> doing wrong here. >> >> _________________________________________________________________________ >> >> 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 > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > 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 piotr at dataandsignal.com Sun Dec 29 23:45:43 2019 From: piotr at dataandsignal.com (Piotr Gregor) Date: Sun, 29 Dec 2019 23:45:43 +0000 Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: Hi Nathan, The best option is to upgrade system to one that ships with SSL above 1.1.0, like for instance new Debian 10 "Buster". peter at photon:~/$ openssl version OpenSSL 1.1.1d 10 Sep 2019 cheers, Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Thu, Dec 12, 2019 at 4:03 PM kaiduan xie via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kaiduan xie > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 15:31:40 +0000 (UTC) > Subject: Re: [Freeswitch-users] Upgrading DTLS > Looks like your SSL version is old. > > #if OPENSSL_VERSION_NUMBER >= 0x10100000 > > https://github.com/signalwire/freeswitch/blob/master/src/switch_rtp.c#L3757 > > The following is the SSL version on 16.0.4 Ubuntu. > > *~*:*/usr/include/openssl*$ grep -R VERSION_NUMBER . -n > > ./crypto.h:152:# define SSLEAY_*VERSION_NUMBER* OPENSSL_*VERSION_NUMBER* > > ./ssl.h:2868:# define SSL_R_BAD_PROTOCOL_*VERSION_NUMBER* > 116 > > ./ssl.h:3164:# define SSL_R_WRONG_*VERSION_NUMBER* > 267 > > ./pem.h:589:# define PEM_R_BAD_*VERSION_NUMBER* > 117 > > ./opensslv.h:33:# define OPENSSL_*VERSION_NUMBER* 0x1000207fL > > ./opensslv.h:83: * The current library version is stored in the macro > SHLIB_*VERSION_NUMBER*, > > ./opensslv.h:91:# define SHLIB_*VERSION_NUMBER* "1.0.0" > > :*/usr/include/openssl*$ lsb_release -a > > No LSB modules are available. > > Distributor ID: Ubuntu > > Description: Ubuntu 16.04.5 LTS > > Release: 16.04 > > Codename: xenial > > > > On Thursday, December 12, 2019, 09:00:42 a.m. CST, Mirko Brankovic < > mirkobrankovic at gmail.com> wrote: > > > VERSION="16.04.6 LTS (Xenial Xerus)" > ~# dpkg -l | grep openssl > ii libcurl4-openssl-dev:amd64 7.47.0-1ubuntu2.14 > amd64 development files and documentation for > libcurl (OpenSSL flavour) > ii libgnutls-openssl27:amd64 3.4.10-4ubuntu1.5 > amd64 GNU TLS library - OpenSSL wrapper > ii libxmlsec1-openssl 1.2.20-2ubuntu4 > amd64 Openssl engine for the XML security library > ii openssl 1.0.2g-1ubuntu4.15 > amd64 Secure Sockets Layer toolkit - > cryptographic utility > > > But the real problem appeared on another webrtc gateway (Janus) that > required TLS 1.2 minimum > > On Thu, Dec 12, 2019 at 3:48 PM kaiduan xie via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > > > > > ---------- Forwarded message ---------- > From: kaiduan xie > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 14:47:41 +0000 (UTC) > Subject: Re: [Freeswitch-users] Upgrading DTLS > What OS and version you run FS on? What is the openssl version on the box? > > /Kaiduan > > On Thursday, December 12, 2019, 03:29:32 a.m. CST, Mirko Brankovic < > mirkobrankovic at gmail.com> wrote: > > > I had a same problem, and I see you can set it in vars.conf: > > https://github.com/signalwire/freeswitch/blob/master/conf/vanilla/vars.xml#L407 > but since we have a custom module, it didn't work for me, so I replaced > OpenSSL with BorringSSL and fixed it that way :D > > On Wed, Dec 11, 2019 at 10:05 PM Nathan Stratton > wrote: > > > Seeing this error on FreeSWITCH 1.10.1 > > 2019-12-11 00:19:34.288375 [ERR] switch_rtp.c:3266 video Handshake failure > 1. This may happen when you use legacy DTLS v1.0 (legacyDTLS channel var is > set) but endpoint requires DTLS v1.2. > > Any idea how to upgrade DTLS to 1.2? I could not find much with a google > search. > > ><> > nathan stratton > _________________________________________________________________________ > > 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 > > > > -- > Regards, > Mirko > _________________________________________________________________________ > > 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 > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 06:48:21 -0800 (PST) > Subject: Re: [Freeswitch-users] Upgrading DTLS > _________________________________________________________________________ > > 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 > > > > -- > Regards, > Mirko > _________________________________________________________________________ > > 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 > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Thu, 12 Dec 2019 08:03:24 -0800 (PST) > Subject: Re: [Freeswitch-users] Upgrading DTLS > _________________________________________________________________________ > > 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 tanim at surroundapps.com Mon Dec 30 13:15:06 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Mon, 30 Dec 2019 19:15:06 +0600 Subject: [Freeswitch-users] No audio for conference call Message-ID: Hi I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper clients. When I call the conference number, I can hear hold music. when the second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this? Audio is fine when I call other users directly. Regards Tanim -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Mon Dec 30 13:51:00 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Mon, 30 Dec 2019 16:51:00 +0300 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Is call disconnected after 34 second? On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi > > I have tried to test FreeSWITCH conference by dialing 3000 from two zoiper > clients. When I call the conference number, I can hear hold music. when > the second call dial the conference number hold music stops, but no voice > between legs. what would be the reason for this? > > Audio is fine when I call other users directly. > > Regards > Tanim > _________________________________________________________________________ > > 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 nathan at robotics.net Mon Dec 30 15:12:00 2019 From: nathan at robotics.net (Nathan Stratton) Date: Mon, 30 Dec 2019 10:12:00 -0500 Subject: [Freeswitch-users] Upgrading DTLS In-Reply-To: References: Message-ID: Yes, thanks, the issue was Centos 7.x does not uspport the correct version, I had to move to Centos 8. ><> nathan stratton On Sun, Dec 29, 2019 at 6:47 PM Piotr Gregor wrote: > Hi Nathan, > > The best option is to upgrade system to one that ships with SSL above > 1.1.0, like for instance new Debian 10 "Buster". > > peter at photon:~/$ openssl version > OpenSSL 1.1.1d 10 Sep 2019 > > cheers, > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathan at robotics.net Mon Dec 30 17:28:10 2019 From: nathan at robotics.net (Nathan Stratton) Date: Mon, 30 Dec 2019 12:28:10 -0500 Subject: [Freeswitch-users] TLS1.2 ONLY Message-ID: # vars.xml # internal.xml However when I check out my server with https://www.ssllabs.com/ssltest/analyze.html it shows TLS1.1 along with TLS1.2. Any way to JUST have the server answer TLS1.2? Also is it possible to set the order of cipher suites and get rid of: Cipher Suites # TLS 1.2 (server has no preference) TLS_RSA_WITH_AES_128_CBC_SHA (0x2f) *WEAK* 128 TLS_RSA_WITH_CAMELLIA_128_CBC_SHA (0x41) *WEAK* 128 TLS_RSA_WITH_AES_128_CBC_SHA256 (0x3c) *WEAK* 128 TLS_RSA_WITH_AES_128_GCM_SHA256 (0x9c) *WEAK* 128 TLS_RSA_WITH_AES_256_CBC_SHA (0x35) *WEAK* 256 TLS_RSA_WITH_CAMELLIA_256_CBC_SHA (0x84) *WEAK* 256 TLS_RSA_WITH_AES_256_CBC_SHA256 (0x3d) *WEAK* 256 TLS_RSA_WITH_AES_256_GCM_SHA384 (0x9d) *WEAK* 256 ><> nathan stratton -------------- next part -------------- An HTML attachment was scrubbed... URL: From napole at gmail.com Sun Dec 29 16:57:33 2019 From: napole at gmail.com (Mitchell Langs) Date: Sun, 29 Dec 2019 17:57:33 +0100 Subject: [Freeswitch-users] uuid_simplify after intercept/bridge Message-ID: Hi, I need to originate 2 calls and later bridge them. Then Freeswitch needs to be removed from the call path. uuid_simplify accomplishes this. But where do I put this command so that it is executed once the two calls are bridged? As far as I understand the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge" command until the bridge is terminated. I have also tried to set sip_auto_simplify=true but it had no effect. Here's what I am executing: originate {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620 testing XML public ... Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Mon Dec 30 19:47:27 2019 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 30 Dec 2019 12:47:27 -0700 Subject: [Freeswitch-users] uuid_simplify after intercept/bridge In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge > On Dec 29, 2019, at 9:57 AM, Mitchell Langs wrote: > > Hi, > I need to originate 2 calls and later bridge them. Then Freeswitch needs to be removed from the call path. > > uuid_simplify accomplishes this. But where do I put this command so that it is executed once the two calls are bridged? As far as I understand the dialplan or scripts pause when I issue the "intercept" or "uuid_bridge" command until the bridge is terminated. > > I have also tried to set sip_auto_simplify=true but it had no effect. > > > Here's what I am executing: > > originate {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620 testing XML public > > > > > ... > > > > > Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From tanim at surroundapps.com Tue Dec 31 03:56:06 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Tue, 31 Dec 2019 09:56:06 +0600 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Hi Sergey No, call is not disconnected. Only some noise can be heard after enabling speaker. Regards Tanim On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov wrote: > Is call disconnected after 34 second? > > On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < > tanim at surroundapps.com> wrote: > >> Hi >> >> I have tried to test FreeSWITCH conference by dialing 3000 from two >> zoiper clients. When I call the conference number, I can hear hold >> music. when the second call dial the conference number hold music stops, >> but no voice between legs. what would be the reason for this? >> >> Audio is fine when I call other users directly. >> >> Regards >> Tanim >> _________________________________________________________________________ >> >> 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 s.safarov at gmail.com Tue Dec 31 05:22:35 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 31 Dec 2019 08:22:35 +0300 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: may be users connected with "mute" or "deaf" option. On Tue, Dec 31, 2019 at 7:28 AM Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi Sergey > > No, call is not disconnected. Only some noise can be heard after enabling > speaker. > > Regards > Tanim > > On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov > wrote: > >> Is call disconnected after 34 second? >> >> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >> tanim at surroundapps.com> wrote: >> >>> Hi >>> >>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>> zoiper clients. When I call the conference number, I can hear hold >>> music. when the second call dial the conference number hold music >>> stops, but no voice between legs. what would be the reason for this? >>> >>> Audio is fine when I call other users directly. >>> >>> Regards >>> Tanim >>> _________________________________________________________________________ >>> >>> 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 imfanee at gmail.com Tue Dec 31 06:42:41 2019 From: imfanee at gmail.com (Faisal Hanif) Date: Tue, 31 Dec 2019 11:42:41 +0500 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Seems a complex NAT issue. Share the detailed call flow. On Tue, 31 Dec 2019, 9:28 am Md,Mehedi Hasan Kabir(Tanim), < tanim at surroundapps.com> wrote: > Hi Sergey > > No, call is not disconnected. Only some noise can be heard after enabling > speaker. > > Regards > Tanim > > On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov > wrote: > >> Is call disconnected after 34 second? >> >> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >> tanim at surroundapps.com> wrote: >> >>> Hi >>> >>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>> zoiper clients. When I call the conference number, I can hear hold >>> music. when the second call dial the conference number hold music >>> stops, but no voice between legs. what would be the reason for this? >>> >>> Audio is fine when I call other users directly. >>> >>> Regards >>> Tanim >>> _________________________________________________________________________ >>> >>> 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 s.safarov at gmail.com Tue Dec 31 08:54:02 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 31 Dec 2019 11:54:02 +0300 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: If NAT issue, then FreeSwitch not able to receive audio from phone. If FreeSwitch not able receive audio, then call will be disconnected FreeSwitch side. On Tue, Dec 31, 2019, 9:58 AM Faisal Hanif wrote: > Seems a complex NAT issue. Share the detailed call flow. > > On Tue, 31 Dec 2019, 9:28 am Md,Mehedi Hasan Kabir(Tanim), < > tanim at surroundapps.com> wrote: > >> Hi Sergey >> >> No, call is not disconnected. Only some noise can be heard after enabling >> speaker. >> >> Regards >> Tanim >> >> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov >> wrote: >> >>> Is call disconnected after 34 second? >>> >>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >>> tanim at surroundapps.com> wrote: >>> >>>> Hi >>>> >>>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>>> zoiper clients. When I call the conference number, I can hear hold >>>> music. when the second call dial the conference number hold music >>>> stops, but no voice between legs. what would be the reason for this? >>>> >>>> Audio is fine when I call other users directly. >>>> >>>> Regards >>>> Tanim >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 > > _________________________________________________________________________ > > 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 tanim at surroundapps.com Tue Dec 31 09:03:38 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Tue, 31 Dec 2019 15:03:38 +0600 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Hi Sergey Actually i just dialled 3000 from Zoiper mobile client to test freeswitch default conference whose dialplan is default at freeswitch default.xml file. in cli, conference 3000-dialengine.sensor.buzz json_list command output is as follows [{ "conference_name": "3000-dialengine.sensor.buzz", "member_count": 2, "ghost_count": 0, "rate": 8000, "run_time": 235, "conference_uuid": "af76568b-80e1-4def-8c0a-f2e2b3076925", "canvas_count": 0, "max_bw_in": 0, "force_bw_in": 0, "video_floor_packets": 0, "locked": false, "destruct": false, "wait_mod": false, "audio_always": false, "running": true, "answered": true, "enforce_min": true, "bridge_to": false, "dynamic": true, "exit_sound": true, "enter_sound": true, "recording": false, "video_bridge": false, "video_floor_only": false, "video_rfc4579": false, "variables": { }, "members": [{ "type": "caller", "id": 20, "flags": { "can_hear": true, "can_see": true, "can_speak": true, "hold": false, "mute_detect": false, "talking": false, "has_video": false, "video_bridge": false, "has_floor": false, "is_moderator": false, "end_conference": false }, "uuid": "584d5363-6243-4051-bb3e-6b2365ea4ed6", "caller_id_name": "2112", "caller_id_number": "2112", "join_time": 21, "last_talking": 0, "energy": 100, "volume_in": 0, "volume_out": 0, "output-volume": 0, "input-volume": 0 }, { "type": "caller", "id": 19, "flags": { "can_hear": true, "can_see": true, "can_speak": true, "hold": false, "mute_detect": false, "talking": false, "has_video": false, "video_bridge": false, "has_floor": true, "is_moderator": false, "end_conference": false }, "uuid": "3eb72d21-f767-4c49-a83f-f4d6ddc78eb4", "caller_id_name": "2111", "caller_id_number": "2111", "join_time": 235, "last_talking": 0, "energy": 100, "volume_in": 0, "volume_out": 0, "output-volume": 0, "input-volume": 0 }] }] Can you guess anything from this log? On Tue, Dec 31, 2019 at 2:54 PM Sergey Safarov wrote: > If NAT issue, then FreeSwitch not able to receive audio from phone. > If FreeSwitch not able receive audio, then call will be disconnected > FreeSwitch side. > > On Tue, Dec 31, 2019, 9:58 AM Faisal Hanif wrote: > >> Seems a complex NAT issue. Share the detailed call flow. >> >> On Tue, 31 Dec 2019, 9:28 am Md,Mehedi Hasan Kabir(Tanim), < >> tanim at surroundapps.com> wrote: >> >>> Hi Sergey >>> >>> No, call is not disconnected. Only some noise can be heard after >>> enabling speaker. >>> >>> Regards >>> Tanim >>> >>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov >>> wrote: >>> >>>> Is call disconnected after 34 second? >>>> >>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >>>> tanim at surroundapps.com> wrote: >>>> >>>>> Hi >>>>> >>>>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>>>> zoiper clients. When I call the conference number, I can hear hold >>>>> music. when the second call dial the conference number hold music >>>>> stops, but no voice between legs. what would be the reason for this? >>>>> >>>>> Audio is fine when I call other users directly. >>>>> >>>>> Regards >>>>> Tanim >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 >> >> _________________________________________________________________________ >> >> 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 david.villasmil.work at gmail.com Tue Dec 31 10:36:13 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 31 Dec 2019 11:36:13 +0100 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Check the IP addresses offered from Zoiper and freeswitch. Make sure none is private. Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip ? On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi Sergey > > No, call is not disconnected. Only some noise can be heard after enabling > speaker. > > Regards > Tanim > > On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov > wrote: > >> Is call disconnected after 34 second? >> >> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >> tanim at surroundapps.com> wrote: >> >>> Hi >>> >>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>> zoiper clients. When I call the conference number, I can hear hold >>> music. when the second call dial the conference number hold music >>> stops, but no voice between legs. what would be the reason for this? >>> >>> Audio is fine when I call other users directly. >>> >>> Regards >>> Tanim >>> _________________________________________________________________________ >>> >>> 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From napole at gmail.com Tue Dec 31 10:49:51 2019 From: napole at gmail.com (Mitchell Langs) Date: Tue, 31 Dec 2019 11:49:51 +0100 Subject: [Freeswitch-users] uuid_simplify after intercept/bridge In-Reply-To: References: Message-ID: Thank you for the idea, Mike! However bypass_media_after_bridge only handles the media. The sip messages are still being exchanged over Freeswitch after this. That means the calls between freeswitch and the gateway are still active. The gateway has a limit for concurrent calls. Therefore I want to call uuid_simplify when the bridge starts. For the gateway this would reduce the number of concurrent calls by two. Am Mo., 30. Dez. 2019 um 20:57 Uhr schrieb Mike Jerris : > > https://freeswitch.org/confluence/display/FREESWITCH/bypass_media_after_bridge > > On Dec 29, 2019, at 9:57 AM, Mitchell Langs wrote: > > Hi, > I need to originate 2 calls and later bridge them. Then Freeswitch needs > to be removed from the call path. > > uuid_simplify accomplishes this. But where do I put this command so that > it is executed once the two calls are bridged? As far as I understand the > dialplan or scripts pause when I issue the "intercept" or "uuid_bridge" > command until the bridge is terminated. > > I have also tried to set sip_auto_simplify=true but it had no effect. > > > Here's what I am executing: > > originate > {codec_string=PCMA,ignore_early_media=true}sofia/gateway/fritzbox/**620 > testing XML public > > > > > ... > > > > > Kind Regards > > > _________________________________________________________________________ > > 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 babak.freeswitch at gmail.com Tue Dec 31 11:18:18 2019 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Tue, 31 Dec 2019 14:48:18 +0330 Subject: [Freeswitch-users] calling sleep after answer results in no audio Message-ID: Hi trying below dialplan results in no audio: but if I move *sleep before answer* everything works fine. Using tcpdump shows that when sleep is called after answer no rtp is established but if I move sleep before answer rtp is established I'm using freeswitch 1.8.7 on debian 8 thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From tanim at surroundapps.com Tue Dec 31 13:44:59 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Tue, 31 Dec 2019 19:44:59 +0600 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Hi David Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two zoiper client. problem occurs only for conference call. Regards Tanim On Tue, Dec 31, 2019, 4:36 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Check the IP addresses offered from Zoiper and freeswitch. Make sure none > is private. > > Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip ? > > On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) < > tanim at surroundapps.com> wrote: > >> Hi Sergey >> >> No, call is not disconnected. Only some noise can be heard after enabling >> speaker. >> >> Regards >> Tanim >> >> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov >> wrote: >> >>> Is call disconnected after 34 second? >>> >>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >>> tanim at surroundapps.com> wrote: >>> >>>> Hi >>>> >>>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>>> zoiper clients. When I call the conference number, I can hear hold >>>> music. when the second call dial the conference number hold music >>>> stops, but no voice between legs. what would be the reason for this? >>>> >>>> Audio is fine when I call other users directly. >>>> >>>> Regards >>>> Tanim >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 > > -- > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Dec 31 14:33:40 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 31 Dec 2019 15:33:40 +0100 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: can you please share a SIP trace and all FS logs? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Dec 31, 2019 at 2:59 PM Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi David > > Yes,ext-sip-ip and ext-rtp-ip is set.Audio is ok for call between two > zoiper client. problem occurs only for conference call. > > Regards > Tanim > > > On Tue, Dec 31, 2019, 4:36 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Check the IP addresses offered from Zoiper and freeswitch. Make sure none >> is private. >> >> Is fs behind NAT? If so, did you set the ext-sip-ip and ext-rtp-ip ? >> >> On Tue, 31 Dec 2019 at 05:24, Md,Mehedi Hasan Kabir(Tanim) < >> tanim at surroundapps.com> wrote: >> >>> Hi Sergey >>> >>> No, call is not disconnected. Only some noise can be heard after >>> enabling speaker. >>> >>> Regards >>> Tanim >>> >>> On Mon, Dec 30, 2019 at 7:51 PM Sergey Safarov >>> wrote: >>> >>>> Is call disconnected after 34 second? >>>> >>>> On Mon, Dec 30, 2019 at 4:37 PM Md,Mehedi Hasan Kabir(Tanim) < >>>> tanim at surroundapps.com> wrote: >>>> >>>>> Hi >>>>> >>>>> I have tried to test FreeSWITCH conference by dialing 3000 from two >>>>> zoiper clients. When I call the conference number, I can hear hold >>>>> music. when the second call dial the conference number hold music >>>>> stops, but no voice between legs. what would be the reason for this? >>>>> >>>>> Audio is fine when I call other users directly. >>>>> >>>>> Regards >>>>> Tanim >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 >> >> -- >> 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 > > _________________________________________________________________________ > > 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 davidswalkabout at gmail.com Tue Dec 31 15:56:09 2019 From: davidswalkabout at gmail.com (David P) Date: Tue, 31 Dec 2019 10:56:09 -0500 Subject: [Freeswitch-users] No audio for conference call In-Reply-To: References: Message-ID: Search your log for 'Choose rtp'; the IP number following this shows what FS will use for a media stream. There is a line like this for each stream. For your scenario, you want these not to be private IPs. If they are private, you can try to auto-reject them by changing acl.conf.xml. But you'll have limited success due to this bug https://github.com/signalwire/freeswitch/issues/157 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Tue Dec 31 17:47:06 2019 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 31 Dec 2019 10:47:06 -0700 Subject: [Freeswitch-users] calling sleep after answer results in no audio In-Reply-To: References: Message-ID: <20378B1F-DE92-4B5A-9FC2-89E3A5B270F8@freeswitch.org> Is this the case on the most recent release? > On Dec 31, 2019, at 4:18 AM, Babak Yakhchali wrote: > > Hi > trying below dialplan results in no audio: > > > > > > > > > > > > > but if I move sleep before answer everything works fine. Using tcpdump shows that when sleep is called after answer no rtp is established but if I move sleep before answer rtp is established > I'm using freeswitch 1.8.7 on debian 8 > thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From obelousov at gmail.com Tue Dec 31 14:40:03 2019 From: obelousov at gmail.com (Oleg Belousov) Date: Tue, 31 Dec 2019 17:40:03 +0300 Subject: [Freeswitch-users] bypass_media_after_bridge Message-ID: Happy festive season. Would you please guide me with usage of 'bypass_media_after_bridge' feature? Call flow is as following (running in lua script): 1. freeswitch preanswer call from UAC, then play some prompts 2, create second_session 3. when second session is ready - assign bypass_media_after_bridge=true and, see var is assigned. EXECUTE [depth=0] sofia/external/XXXXXXXXXX set(bypass_media_after_bridge=true) 2019-12-31 14:21:19.253279 [DEBUG] mod_dptools.c:1672 SET sofia/external/XXXXXXXXXX [bypass_media_after_bridge]=[true] 4. bridge both calls. At that time expect freeswtch fire re-invite to get out of rtp path, however do not see any re-invites. extract from script: ---- ConnectStr = "sofia/gateway/aed8dd2f-202f-4093-9851-436e21bafcf4/"..DN second_session = freeswitch.Session(ConnectStr) if (second_session:ready()) then second_session:execute("set","bypass_media_after_bridge=true") freeswitch.bridge(session, second_session) --- What could be wrong in my script? version: FreeSWITCH Version 1.10.1-release-12 -- Regards, Oleg -------------- next part -------------- An HTML attachment was scrubbed... URL: From andywolk at gmail.com Tue Dec 31 17:54:22 2019 From: andywolk at gmail.com (Andrey Wolk) Date: Tue, 31 Dec 2019 21:54:22 +0400 Subject: [Freeswitch-users] FreeSWITCH v.1.10.2 Released Message-ID: WooHooo! We are happy to present you the freshest FreeSWITCH v.1.10.2 Release! This release contains ~102 improvements. https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.10.x+Release+notes https://github.com/signalwire/freeswitch/releases/tag/v1.10.2 Happy New Year everyone! -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Tue Dec 31 17:54:46 2019 From: mike at freeswitch.org (Mike Jerris) Date: Tue, 31 Dec 2019 10:54:46 -0700 Subject: [Freeswitch-users] bypass_media_after_bridge In-Reply-To: References: Message-ID: <961B66B1-260E-4B87-912F-A9895BA6B641@freeswitch.org> May not work with freeswitch.bridge. try doing a uuid_bridge and see if it makes any difference. > On Dec 31, 2019, at 7:40 AM, Oleg Belousov wrote: > > Happy festive season. > > Would you please guide me with usage of 'bypass_media_after_bridge' feature? > > Call flow is as following (running in lua script): > 1. freeswitch preanswer call from UAC, then play some prompts > 2, create second_session > 3. when second session is ready - assign bypass_media_after_bridge=true and, see var is assigned. > EXECUTE [depth=0] sofia/external/XXXXXXXXXX set(bypass_media_after_bridge=true) > 2019-12-31 14:21:19.253279 [DEBUG] mod_dptools.c:1672 SET sofia/external/XXXXXXXXXX [bypass_media_after_bridge]=[true] > 4. bridge both calls. At that time expect freeswtch fire re-invite to get out of rtp path, however do not see any re-invites. > > extract from script: > ---- > ConnectStr = "sofia/gateway/aed8dd2f-202f-4093-9851-436e21bafcf4/"..DN > second_session = freeswitch.Session(ConnectStr) > > if (second_session:ready()) then > second_session:execute("set","bypass_media_after_bridge=true") > freeswitch.bridge(session, second_session) > --- > > What could be wrong in my script? > version: FreeSWITCH Version 1.10.1-release-12 -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Dec 31 18:50:32 2019 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 31 Dec 2019 21:50:32 +0300 Subject: [Freeswitch-users] calling sleep after answer results in no audio In-Reply-To: <20378B1F-DE92-4B5A-9FC2-89E3A5B270F8@freeswitch.org> References: <20378B1F-DE92-4B5A-9FC2-89E3A5B270F8@freeswitch.org> Message-ID: I tested this case on 1.4 branch http://lists.freeswitch.org/pipermail/freeswitch-users/2015-July/114358.html I able to receive RTP stream after answer. Looks as need make "git bisect" Sergey On Tue, Dec 31, 2019 at 9:11 PM Mike Jerris wrote: > Is this the case on the most recent release? > > On Dec 31, 2019, at 4:18 AM, Babak Yakhchali > wrote: > > Hi > trying below dialplan results in no audio: > uuid="d8037520-2a15-11ea-9a9f-9bf4580ab5b8"> > > > > > > > data="ivr_menu_uuid=d8049250-2a15-11ea-9f14-1b8cf3ef5a32"/> > > > > > but if I move *sleep before answer* everything works fine. Using tcpdump > shows that when sleep is called after answer no rtp is established but if I > move sleep before answer rtp is established > I'm using freeswitch 1.8.7 on debian 8 > 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... 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