From spencer.angerbauer at gmail.com Sat Feb 1 06:27:25 2020 From: spencer.angerbauer at gmail.com (Spencer Angerbauer) Date: Fri, 31 Jan 2020 23:27:25 -0700 Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences + announcement In-Reply-To: <38f2a340-1bef-693e-a339-09b645ff22f2@mst.edu> References: <45214DC2-EBE2-4D6F-8C0F-BD7EF556D1E6@ventureslopes.com> <7E0EFC95-8F9C-4009-A0B7-10B224049FC2@gmail.com> <83DE024C-2765-4E87-84FD-CBA5F8479A3D@gmail.com> <437549DE-925D-45ED-A370-B2CB681C56C9@gmail.com> <2183e072-51ea-e9fb-c638-e6572e7793c1@mst.edu> <38f2a340-1bef-693e-a339-09b645ff22f2@mst.edu> Message-ID: <273D9355-36E1-4B89-813E-85574A000CFA@gmail.com> Thanks guys. The conference_enforce_security=true was the issue. I’m able to add a pin to the originate command line and webapi now with that entry. > On Jan 31, 2020, at 6:06 AM, Nathan Neulinger wrote: > > See if below helps if you add it to the originate webapi call: (You'll need to encode it for query parameter usage.) > > https://freeswitch.org/confluence/display/FREESWITCH/mod_conference > > --------------------------- > conference_enforce_security > Allows the conference security to be overridden. This applies in two different scenarios, one for inbound and one for outbound. > > By default, conference security is always applied to inbound calls and is always skipped for outbound calls. > > Inbound > > > > Outbound > > originate {conference_enforce_security=true}sofia/internal/1001 &conference(3000) > ------------------------- > > > > > From: David Villasmil [mailto:david.villasmil.work at gmail.com ] > Sent: Friday, January 31, 2020, 4:02 AM > To: FreeSWITCH Users Help > Cc: Nathan Neulinger > Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences + announcement > >> Does that api request treats & as fs or as a variable separator? I wonder... never used the web api, though. >> >> I think you’d be better off using ESL. >> >> Regards, >> >> David >> >> On Fri, 31 Jan 2020 at 08:48, Spencer Angerbauer > wrote: >> Thanks Nathan for the quick reply. We are creating our conferences through the webapi and triggering a web request for each number joining a particular call with the same conference ID… so for example, if we were bridging 3 calls, we are sending a GET post to the API sequentially by the following example phone numbers: >> >> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/15555555555%20&conference(1234) >> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/17777777777%20&conference(1234) >> http://12.34.56.78:8080/webapi/originate?sofia/gateway/signalwire/18888888888%20&conference(1234) >> >> >> We’ve tried to pass the pin via the webapi post without luck, and as mentioned below have tried to even hard-set the xml in conference.conf.xml to hardcoded to “1” as the PIN with no luck. >> >> Is there a way to require the pin from the Webapi example above or even from the originate example here?: >> >> originate sofia/gateway/signalwire/15555555555 &conference(1234) >> >> Thank you again Nathan for your quick response today! >> >> -Spence >> >> >>> On Jan 30, 2020, at 7:02 PM, Nathan Neulinger > wrote: >>> >>> How are you creating your conference? >>> >>> If you have that pin setting in the conference profile named "mysettings", you should be able to create your conference using the "conference" application specifying that profile. >>> >>> i.e. >>> >>> >>> >>> or you can just do it on that create >>> >>> >>> >>> >>> See docs here: https://freeswitch.org/confluence/display/FREESWITCH/mod_conference >>> >>> -- Nathan >>> >>> From: Spencer Angerbauer [mailto:spencer.angerbauer at gmail.com ] >>> Sent: Thursday, January 30, 2020, 6:33 PM >>> To: FreeSWITCH Users Help >>> Subject: [Freeswitch-users] Requiring Pin of "1" for all conferences + announcement >>> >>>> Just following up after troubleshooting a variety of items… is there an easier way to hard code the XML to require PIN “1” in the conference.conf.xml files for every and all conference calls on the freeswitch pbx? It seems the issue is requiring a PIN on a web api call or an originate call from the console. I am simply trying to hard set Pins for all conferences to be “1” for testing purposes. Any help would be greatly appreciated. Am I missing an additional required parameter? >>>> >>>> Thanks, >>>> >>>> -Spence >>>> >>>>> On Jan 23, 2020, at 10:28 PM, Spencer Angerbauer > wrote: >>>>> >>>>> Ive also update all conference.con.xml to include the following: >>>>> >>>>> >>>>> >>>>> >> seconds? I thought it was 1 carrier on 1 route, but I'm getting reports now >>> across various destinations using different carriers. It's not on every >>> call, just some. >>> > I've upgraded to FreeSWITCH Version 1.10.2-release-13-f7bdd3845a~64bit >>> (-release-13-f7bdd3845a 64bit) and we'll see if I still get the issue. >>> > >>> > >>> > Thanks, >>> > -Avi Marcus >>> > >>> _________________________________________________________________________ >>> >>> 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 gregor at infomedia.si Sat Feb 1 18:32:26 2020 From: gregor at infomedia.si (Gregor Nanger) Date: Sat, 1 Feb 2020 19:32:26 +0100 Subject: [Freeswitch-users] Calls repeat at 80 seconds? In-Reply-To: References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com> Message-ID: https://en.m.wikipedia.org/wiki/False_Answer_Supervision One repeats audio conversation to keep a side connected as long as possible although b side is disconnected on carrier side. Can you check, if upon audio repetition, end user phone is disconnected? On Sat, 1 Feb 2020, 15:10 Tamas Jalsovszky, wrote: > What does FAS stand for? > Any other idea if not a fraud, what kind of tech issue it might be? I was > thinking about a messed echo canceller or messed up jitterbuffer, but it is > not very probable due to long timing (80 sec is extremely high for such > case). > > On Sun, 12 Jan 2020 at 23:36, Gregor Nanger wrote: > >> It's called FAS 😊 >> >> On Sun, 12 Jan 2020, 19:04 David Villasmil, < >> david.villasmil.work at gmail.com> wrote: >> >>> +1 >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> >>> On Sun, Jan 12, 2020 at 5:01 PM Brian : wrote: >>> >>>> There were some carries that would do this with high value middle >>>> Eastern destinations. Record first x seconds of call then hangup the actual >>>> call to the high value destination and play the recording back. The caller >>>> would stay connected and be billed. The unscrupulous carrier thus getting >>>> high value per minute rates and not paying it on as the call was hung up. >>>> >>>> >>>> On Sunday, January 12, 2020, Avi Marcus wrote: >>>> > FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) >>>> > Is anyone having an issue with calls suddenly "restarting" at about >>>> 80 seconds? I thought it was 1 carrier on 1 route, but I'm getting reports >>>> now across various destinations using different carriers. It's not on every >>>> call, just some. >>>> > I've upgraded to FreeSWITCH Version >>>> 1.10.2-release-13-f7bdd3845a~64bit (-release-13-f7bdd3845a 64bit) and we'll >>>> see if I still get the issue. >>>> > >>>> > >>>> > Thanks, >>>> > -Avi Marcus >>>> > >>>> _________________________________________________________________________ >>>> >>>> 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 william at williamcollsassoc.ca Sat Feb 1 23:00:09 2020 From: william at williamcollsassoc.ca (William Colls) Date: Sat, 1 Feb 2020 18:00:09 -0500 Subject: [Freeswitch-users] Restricting In bound calls Message-ID: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> I am sure that I read somewhere that there is a parameter that i can set so that only incoming calls from specific IP address are accepted, but I can't find the reference, remember the parameter name, or where it is located. Could some kind sole point me in the right direction? Thanks for you time. William. From tom at tomlynn.com Sun Feb 2 03:27:36 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sat, 1 Feb 2020 19:27:36 -0800 Subject: [Freeswitch-users] Restricting In bound calls In-Reply-To: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> References: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> Message-ID: Have you looked in acl.conf.xml located in autoload_configs? On Sat, Feb 1, 2020 at 3:03 PM William Colls wrote: > > I am sure that I read somewhere that there is a parameter that i can set > so that only incoming calls from specific IP address are accepted, but I > can't find the reference, remember the parameter name, or where it is > located. Could some kind sole point me in the right direction? > > Thanks for you time. > > William. > > > _________________________________________________________________________ > > 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 tom at tomlynn.com Sun Feb 2 06:59:51 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sat, 1 Feb 2020 22:59:51 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: <20150911164506.GA8640@hau.nz> References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: For what it's worth, I'm running into this very same issue over 4 years after the original poster. I suspect that leaving off the group option is leaving the process running as group root, which would mimic my attempt at having the service run as user freeswitch/group root, which allows mod_portaudio to function. I've posted here in the mail list, on IRC and in Slack, but the only bite was someone hinting about freeswitch user needing to be in group audio, which it is. This is broken. I've looked at submitting a JIRA on this, but my login doesn't appear to allow it. On Fri, Sep 11, 2015 at 9:45 AM Mark Haun wrote: > covici at ccs.covici.com [covici at ccs.covici.com] wrote: > > If using alsa, did you check the permission of /dev/snd and its > > children? > > Yes. In fact, both alsa operations (aplay) and portaudio enumeration work > correctly when running as the freeswitch user. All of this is detailed in > my initial post, two or three weeks ago. > > Mark > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jalsot at gmail.com Sun Feb 2 07:36:50 2020 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Sun, 2 Feb 2020 08:36:50 +0100 Subject: [Freeswitch-users] Calls repeat at 80 seconds? In-Reply-To: References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com> Message-ID: Thank you for details, we will check for the symptoms (whether end user phone was disconnected earlier or not). If we are not hitting FAS, any tech idea what could cause such an issue? On Sat, 1 Feb 2020 at 20:42, Gregor Nanger wrote: > https://en.m.wikipedia.org/wiki/False_Answer_Supervision > > One repeats audio conversation to keep a side connected as long as > possible although b side is disconnected on carrier side. > > Can you check, if upon audio repetition, end user phone is disconnected? > > > On Sat, 1 Feb 2020, 15:10 Tamas Jalsovszky, wrote: > >> What does FAS stand for? >> Any other idea if not a fraud, what kind of tech issue it might be? I was >> thinking about a messed echo canceller or messed up jitterbuffer, but it is >> not very probable due to long timing (80 sec is extremely high for such >> case). >> >> On Sun, 12 Jan 2020 at 23:36, Gregor Nanger wrote: >> >>> It's called FAS 😊 >>> >>> On Sun, 12 Jan 2020, 19:04 David Villasmil, < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> +1 >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> phone: +34669448337 >>>> >>>> >>>> On Sun, Jan 12, 2020 at 5:01 PM Brian : wrote: >>>> >>>>> There were some carries that would do this with high value middle >>>>> Eastern destinations. Record first x seconds of call then hangup the actual >>>>> call to the high value destination and play the recording back. The caller >>>>> would stay connected and be billed. The unscrupulous carrier thus getting >>>>> high value per minute rates and not paying it on as the call was hung up. >>>>> >>>>> >>>>> On Sunday, January 12, 2020, Avi Marcus wrote: >>>>> > FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) >>>>> > Is anyone having an issue with calls suddenly "restarting" at about >>>>> 80 seconds? I thought it was 1 carrier on 1 route, but I'm getting reports >>>>> now across various destinations using different carriers. It's not on every >>>>> call, just some. >>>>> > I've upgraded to FreeSWITCH Version >>>>> 1.10.2-release-13-f7bdd3845a~64bit (-release-13-f7bdd3845a 64bit) and we'll >>>>> see if I still get the issue. >>>>> > >>>>> > >>>>> > Thanks, >>>>> > -Avi Marcus >>>>> > >>>>> _________________________________________________________________________ >>>>> >>>>> 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 gregor at infomedia.si Sun Feb 2 07:58:34 2020 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 2 Feb 2020 08:58:34 +0100 Subject: [Freeswitch-users] Calls repeat at 80 seconds? In-Reply-To: References: <0100016f99dce3c1-51f0a57f-e4f7-4c52-85e7-fc08c20a713f-000000@email.amazonses.com> Message-ID: If you hear same conversation from start, then it has to be recorded somehow. Doubt it's a delay, especially if first 80 seconds is regular two way audio. Try enable debug and see what is happening during the call and record conversation with wireshark. Or try to upgrade FS. On Sun, 2 Feb 2020, 08:37 Tamas Jalsovszky, wrote: > Thank you for details, we will check for the symptoms (whether end user > phone was disconnected earlier or not). > If we are not hitting FAS, any tech idea what could cause such an issue? > > On Sat, 1 Feb 2020 at 20:42, Gregor Nanger wrote: > >> https://en.m.wikipedia.org/wiki/False_Answer_Supervision >> >> One repeats audio conversation to keep a side connected as long as >> possible although b side is disconnected on carrier side. >> >> Can you check, if upon audio repetition, end user phone is disconnected? >> >> >> On Sat, 1 Feb 2020, 15:10 Tamas Jalsovszky, wrote: >> >>> What does FAS stand for? >>> Any other idea if not a fraud, what kind of tech issue it might be? I >>> was thinking about a messed echo canceller or messed up jitterbuffer, but >>> it is not very probable due to long timing (80 sec is extremely high for >>> such case). >>> >>> On Sun, 12 Jan 2020 at 23:36, Gregor Nanger wrote: >>> >>>> It's called FAS 😊 >>>> >>>> On Sun, 12 Jan 2020, 19:04 David Villasmil, < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> +1 >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Sun, Jan 12, 2020 at 5:01 PM Brian : wrote: >>>>> >>>>>> There were some carries that would do this with high value middle >>>>>> Eastern destinations. Record first x seconds of call then hangup the actual >>>>>> call to the high value destination and play the recording back. The caller >>>>>> would stay connected and be billed. The unscrupulous carrier thus getting >>>>>> high value per minute rates and not paying it on as the call was hung up. >>>>>> >>>>>> >>>>>> On Sunday, January 12, 2020, Avi Marcus wrote: >>>>>> > FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) >>>>>> > Is anyone having an issue with calls suddenly "restarting" at about >>>>>> 80 seconds? I thought it was 1 carrier on 1 route, but I'm getting reports >>>>>> now across various destinations using different carriers. It's not on every >>>>>> call, just some. >>>>>> > I've upgraded to FreeSWITCH Version >>>>>> 1.10.2-release-13-f7bdd3845a~64bit (-release-13-f7bdd3845a 64bit) and we'll >>>>>> see if I still get the issue. >>>>>> > >>>>>> > >>>>>> > Thanks, >>>>>> > -Avi Marcus >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> >>>>>> 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 > > _________________________________________________________________________ > > 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 gregor at infomedia.si Sun Feb 2 08:06:22 2020 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 2 Feb 2020 09:06:22 +0100 Subject: [Freeswitch-users] Angular Vero Message-ID: Hi! Anyone ported angularjs verto service to Angular? Best regards, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: From imfanee at gmail.com Sun Feb 2 09:05:35 2020 From: imfanee at gmail.com (Faisal Hanif) Date: Sun, 2 Feb 2020 14:05:35 +0500 Subject: [Freeswitch-users] Angular Vero In-Reply-To: References: Message-ID: Verto calling implimentation on Android worked successfuly. On Sun, Feb 2, 2020, 1:40 PM Gregor Nanger wrote: > Hi! > > Anyone ported angularjs verto service to Angular? > > Best regards, Gregor > _________________________________________________________________________ > > 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 gregor at infomedia.si Sun Feb 2 09:11:10 2020 From: gregor at infomedia.si (Gregor Nanger) Date: Sun, 2 Feb 2020 10:11:10 +0100 Subject: [Freeswitch-users] Angular Vero In-Reply-To: References: Message-ID: Can you please give me more information? On Sun, 2 Feb 2020, 10:06 Faisal Hanif, wrote: > Verto calling implimentation on Android worked successfuly. > > On Sun, Feb 2, 2020, 1:40 PM Gregor Nanger wrote: > >> Hi! >> >> Anyone ported angularjs verto service to Angular? >> >> Best regards, Gregor >> _________________________________________________________________________ >> >> 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 th982a at googlemail.com Sun Feb 2 14:22:31 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 2 Feb 2020 15:22:31 +0100 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: <50dfe544-3b17-3b5b-0bde-c1e05ddc7f12@googlemail.com> Dear Mark, I tell you exactly why. That is less a permission issue. if you open the shell as "root" and type alsamixer you won't be capable to access the soundcard because "root" is not in the 1st) sound group and 2nd) root ha no access to the pulseaudio server This is why I MUST run freeswitch as the user that is currently logged in. mod_alsa doesn't work at all. I didn't get it running, only mod_portaudio.... best, Tamer On 2020-02-02 07:59, Tom Lynn wrote: > For what it's worth, I'm running into this very same issue over 4 > years after the original poster.  I suspect that leaving off the group > option is leaving the process running as group root, which would mimic > my attempt at having the service run as user freeswitch/group root, > which allows mod_portaudio to function. > > I've posted here in the mail list, on IRC and in Slack, but the only > bite was someone hinting about freeswitch user needing to be in group > audio, which it is.  This is broken. I've looked at submitting a JIRA > on this, but my login doesn't appear to allow it. > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > wrote: > > covici at ccs.covici.com > [covici at ccs.covici.com ] wrote: > > If  using alsa, did you check the permission of /dev/snd and its > > children? > > Yes.  In fact, both alsa operations (aplay) and portaudio > enumeration work > correctly when running as the freeswitch user.  All of this is > detailed in > my initial post, two or three weeks ago. > > Mark > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.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 > http://www.freeswitch.org > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com From tom at tomlynn.com Sun Feb 2 16:01:49 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sun, 2 Feb 2020 08:01:49 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: I do not have pulse audio installed. I can access alsamixer and aplay from the shell while logged in as either root or freeswitch without issues. mod_portaudio only works WHEN I'm running freeswitch under group root, which I will not be able to do in a production environment. On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 2 Feb 2020 15:22:31 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > Dear Mark, > > I tell you exactly why. > > That is less a permission issue. > > if you open the shell as "root" and type alsamixer you won't be capable > to access the soundcard because "root" is not in the 1st) sound group > and 2nd) root ha no access to the pulseaudio server > > This is why I MUST run freeswitch as the user that is currently logged in. > > mod_alsa doesn't work at all. I didn't get it running, only > mod_portaudio.... > > > best, Tamer > > > On 2020-02-02 07:59, Tom Lynn wrote: > > For what it's worth, I'm running into this very same issue over 4 > > years after the original poster. I suspect that leaving off the group > > option is leaving the process running as group root, which would mimic > > my attempt at having the service run as user freeswitch/group root, > > which allows mod_portaudio to function. > > > > I've posted here in the mail list, on IRC and in Slack, but the only > > bite was someone hinting about freeswitch user needing to be in group > > audio, which it is. This is broken. I've looked at submitting a JIRA > > on this, but my login doesn't appear to allow it. > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > wrote: > > > > covici at ccs.covici.com > > [covici at ccs.covici.com ] wrote: > > > If using alsa, did you check the permission of /dev/snd and its > > > children? > > > > Yes. In fact, both alsa operations (aplay) and portaudio > > enumeration work > > correctly when running as the freeswitch user. All of this is > > detailed in > > my initial post, two or three weeks ago. > > > > Mark > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales at freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > _________________________________________________________________________ > > 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 th982a at googlemail.com Sun Feb 2 17:02:09 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 2 Feb 2020 18:02:09 +0100 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: <35d9175a-053d-c284-ec1d-4fcf2302a837@googlemail.com> Dear Tom, This is not true. I am running freeswitch as the user I am logged in. as root: chown -R tamer:users /opt/freeswitch as user: /opt/freeswitch/bin/freeswitch -nc as user: fs_cli -r and you will see portaudio runs without any problems. I have it in combination with pulseaudio running. best, Tamer On 2020-02-02 17:01, Tom Lynn wrote: > I do not have pulse audio installed. > > I can access alsamixer and aplay from the shell while logged in as > either root or freeswitch without issues. > > mod_portaudio only works WHEN I'm running freeswitch under group root, > which I will not be able to do in a production environment. > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users > > wrote: > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Sun, 2 Feb 2020 15:22:31 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > Dear Mark, > > I tell you exactly why. > > That is less a permission issue. > > if you open the shell as "root" and type alsamixer you won't be > capable > to access the soundcard because "root" is not in the 1st) sound group > and 2nd) root ha no access to the pulseaudio server > > This is why I MUST run freeswitch as the user that is currently > logged in. > > mod_alsa doesn't work at all. I didn't get it running, only > mod_portaudio.... > > > best, Tamer > > > On 2020-02-02 07:59, Tom Lynn wrote: > > For what it's worth, I'm running into this very same issue over 4 > > years after the original poster.  I suspect that leaving off the > group > > option is leaving the process running as group root, which would > mimic > > my attempt at having the service run as user freeswitch/group root, > > which allows mod_portaudio to function. > > > > I've posted here in the mail list, on IRC and in Slack, but the > only > > bite was someone hinting about freeswitch user needing to be in > group > > audio, which it is.  This is broken. I've looked at submitting a > JIRA > > on this, but my login doesn't appear to allow it. > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > >> wrote: > > > > covici at ccs.covici.com > > > >     [covici at ccs.covici.com > >] wrote: > >     > If  using alsa, did you check the permission of /dev/snd > and its > >     > children? > > > >     Yes.  In fact, both alsa operations (aplay) and portaudio > >     enumeration work > >     correctly when running as the freeswitch user.  All of this is > >     detailed in > >     my initial post, two or three weeks ago. > > > >     Mark > > > > >  _________________________________________________________________________ > >     Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > >     Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.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 > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and > PSTN services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales at freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users > > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > _________________________________________________________________________ > > 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 tom at tomlynn.com Mon Feb 3 00:33:55 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sun, 2 Feb 2020 16:33:55 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: I followed your steps, and yes, pa commands function when freeswitch is run from the command line. Nice! I tried running it as both root and as user tom. Both times I was able to use the pa commands with success. I tried running user tom group user from the systemd startup and was not successful using pa commands. The question becomes, why does this not work using the freeswitch.service configuration file supplied with the packaged freeswitch? I installed freeswitch-meta-all. Tom On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 2 Feb 2020 18:02:09 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > Dear Tom, > > This is not true. > I am running freeswitch as the user I am logged in. > > as root: > chown -R tamer:users /opt/freeswitch > > as user: > /opt/freeswitch/bin/freeswitch -nc > > as user: > fs_cli -r > > and you will see portaudio runs without any problems. > > I have it in combination with pulseaudio running. > > > best, Tamer > > On 2020-02-02 17:01, Tom Lynn wrote: > > I do not have pulse audio installed. > > > > I can access alsamixer and aplay from the shell while logged in as > > either root or freeswitch without issues. > > > > mod_portaudio only works WHEN I'm running freeswitch under group root, > > which I will not be able to do in a production environment. > > > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users > > > > wrote: > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi > > > > To: freeswitch-users at lists.freeswitch.org > > > > Cc: > > Bcc: > > Date: Sun, 2 Feb 2020 15:22:31 +0100 > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > Dear Mark, > > > > I tell you exactly why. > > > > That is less a permission issue. > > > > if you open the shell as "root" and type alsamixer you won't be > > capable > > to access the soundcard because "root" is not in the 1st) sound group > > and 2nd) root ha no access to the pulseaudio server > > > > This is why I MUST run freeswitch as the user that is currently > > logged in. > > > > mod_alsa doesn't work at all. I didn't get it running, only > > mod_portaudio.... > > > > > > best, Tamer > > > > > > On 2020-02-02 07:59, Tom Lynn wrote: > > > For what it's worth, I'm running into this very same issue over 4 > > > years after the original poster. I suspect that leaving off the > > group > > > option is leaving the process running as group root, which would > > mimic > > > my attempt at having the service run as user freeswitch/group root, > > > which allows mod_portaudio to function. > > > > > > I've posted here in the mail list, on IRC and in Slack, but the > > only > > > bite was someone hinting about freeswitch user needing to be in > > group > > > audio, which it is. This is broken. I've looked at submitting a > > JIRA > > > on this, but my login doesn't appear to allow it. > > > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > > > >> wrote: > > > > > > covici at ccs.covici.com > > > > > > [covici at ccs.covici.com > > >] > wrote: > > > > If using alsa, did you check the permission of /dev/snd > > and its > > > > children? > > > > > > Yes. In fact, both alsa operations (aplay) and portaudio > > > enumeration work > > > correctly when running as the freeswitch user. All of this is > > > detailed in > > > my initial post, two or three weeks ago. > > > > > > Mark > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > >> > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.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 > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________________ > > > > > > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > > > Enhance your FreeSWITCH install with disruptive priced SMS and > > PSTN services. > > > Build your next product on our scalable cloud platform. > > > > > > Join our online community to chat in real time > > https://signalwire.community > > > > > > Professional FreeSWITCH Services > > > sales at freeswitch.com > > > https://freeswitch.com > > > > > > Official FreeSWITCH Sites > > > https://freeswitch.com/oss > > > https://freeswitch.org/confluence > > > https://cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > https://freeswitch.com > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi via FreeSWITCH-users > > > > > > To: freeswitch-users at lists.freeswitch.org > > > > Cc: > > Bcc: > > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > > _________________________________________________________________________ > > > > 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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > _________________________________________________________________________ > > 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 tom at tomlynn.com Mon Feb 3 00:43:50 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sun, 2 Feb 2020 16:43:50 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: One more item, I still don't have pulseaudio installed and the pa commands work, so pulseaudio is not strictly necessary. On Sun, Feb 2, 2020 at 4:33 PM Tom Lynn wrote: > I followed your steps, and yes, pa commands function when freeswitch is > run from the command line. Nice! > > I tried running it as both root and as user tom. Both times I was able to > use the pa commands with success. I tried running user tom group user from > the systemd startup and was not successful using pa commands. > > The question becomes, why does this not work using the freeswitch.service > configuration file supplied with the packaged freeswitch? I installed > freeswitch-meta-all. > > Tom > > On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: Tamer Higazi >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Sun, 2 Feb 2020 18:02:09 +0100 >> Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue >> Dear Tom, >> >> This is not true. >> I am running freeswitch as the user I am logged in. >> >> as root: >> chown -R tamer:users /opt/freeswitch >> >> as user: >> /opt/freeswitch/bin/freeswitch -nc >> >> as user: >> fs_cli -r >> >> and you will see portaudio runs without any problems. >> >> I have it in combination with pulseaudio running. >> >> >> best, Tamer >> >> On 2020-02-02 17:01, Tom Lynn wrote: >> > I do not have pulse audio installed. >> > >> > I can access alsamixer and aplay from the shell while logged in as >> > either root or freeswitch without issues. >> > >> > mod_portaudio only works WHEN I'm running freeswitch under group root, >> > which I will not be able to do in a production environment. >> > >> > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via FreeSWITCH-users >> > > > > wrote: >> > >> > >> > >> > >> > ---------- Forwarded message ---------- >> > From: Tamer Higazi > > > >> > To: freeswitch-users at lists.freeswitch.org >> > >> > Cc: >> > Bcc: >> > Date: Sun, 2 Feb 2020 15:22:31 +0100 >> > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > permissions issue >> > Dear Mark, >> > >> > I tell you exactly why. >> > >> > That is less a permission issue. >> > >> > if you open the shell as "root" and type alsamixer you won't be >> > capable >> > to access the soundcard because "root" is not in the 1st) sound >> group >> > and 2nd) root ha no access to the pulseaudio server >> > >> > This is why I MUST run freeswitch as the user that is currently >> > logged in. >> > >> > mod_alsa doesn't work at all. I didn't get it running, only >> > mod_portaudio.... >> > >> > >> > best, Tamer >> > >> > >> > On 2020-02-02 07:59, Tom Lynn wrote: >> > > For what it's worth, I'm running into this very same issue over 4 >> > > years after the original poster. I suspect that leaving off the >> > group >> > > option is leaving the process running as group root, which would >> > mimic >> > > my attempt at having the service run as user freeswitch/group >> root, >> > > which allows mod_portaudio to function. >> > > >> > > I've posted here in the mail list, on IRC and in Slack, but the >> > only >> > > bite was someone hinting about freeswitch user needing to be in >> > group >> > > audio, which it is. This is broken. I've looked at submitting a >> > JIRA >> > > on this, but my login doesn't appear to allow it. >> > > >> > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > >> > > >> wrote: >> > > >> > > covici at ccs.covici.com >> > > >> > > [covici at ccs.covici.com >> > >] >> wrote: >> > > > If using alsa, did you check the permission of /dev/snd >> > and its >> > > > children? >> > > >> > > Yes. In fact, both alsa operations (aplay) and portaudio >> > > enumeration work >> > > correctly when running as the freeswitch user. All of this is >> > > detailed in >> > > my initial post, two or three weeks ago. >> > > >> > > Mark >> > > >> > > >> > >> _________________________________________________________________________ >> > > Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > >> >> > > http://www.freeswitchsolutions.com >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://confluence.freeswitch.org >> > > http://www.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 >> > > http://www.freeswitch.org >> > > >> > > >> > > >> > >> _________________________________________________________________________ >> > > >> > > The FreeSWITCH project is sponsored by SignalWire >> > https://signalwire.com >> > > Enhance your FreeSWITCH install with disruptive priced SMS and >> > PSTN services. >> > > Build your next product on our scalable cloud platform. >> > > >> > > Join our online community to chat in real time >> > https://signalwire.community >> > > >> > > Professional FreeSWITCH Services >> > > sales at freeswitch.com >> > > https://freeswitch.com >> > > >> > > Official FreeSWITCH Sites >> > > https://freeswitch.com/oss >> > > https://freeswitch.org/confluence >> > > https://cluecon.com >> > > >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > https://freeswitch.com >> > >> > >> > >> > >> > ---------- Forwarded message ---------- >> > From: Tamer Higazi via FreeSWITCH-users >> > > > > >> > To: freeswitch-users at lists.freeswitch.org >> > >> > Cc: >> > Bcc: >> > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) >> > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > permissions issue >> > >> _________________________________________________________________________ >> > >> > 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 >> >> >> >> >> ---------- Forwarded message ---------- >> From: Tamer Higazi via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) >> Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue >> _________________________________________________________________________ >> >> 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 th982a at googlemail.com Mon Feb 3 03:57:26 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Mon, 3 Feb 2020 04:57:26 +0100 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: <6a3d09ab-7339-3904-7bc5-ff8184b6e50d@googlemail.com> Dear Tom, It doesn't matter. It works, it shows you all devices. If you can run as usual user "alsamixer" and be capable to list the soundcard and you run freeswitch where all files are owned by this user, and start freeswitch as well as this user then you can access your soundcard with mod_portaudio without any problems. here is my unit file: [Unit] Description=freeswitch After=syslog.target network.target local-fs.target [Service] User=tamer Group=users Type=forking PIDFile=/opt/freeswitch/var/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/opt/freeswitch/bin/freeswitch -nc TimeoutSec=45s Restart=on-failure ; exec WorkingDirectory=/opt/freeswitch/bin LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] WantedBy=multi-user.target here: tamer at tux ~ $ fs_cli -r freeswitch at tux> pa dev [             devlist] freeswitch at tux> pa devlist 0;HDA NVidia: HDMI 0 (hw:0,3)(ALSA);0;8; 1;HDA NVidia: HDMI 1 (hw:0,7)(ALSA);0;8; 2;HDA NVidia: HDMI 2 (hw:0,8)(ALSA);0;2; 3;HDA NVidia: HDMI 3 (hw:0,9)(ALSA);0;8; 4;HD-Audio Generic: ALC892 Analog (hw:1,0)(ALSA);2;2; 5;Logitech Webcam C925e: USB Audio (hw:2,0)(ALSA);2;0; 6;Sound Blaster E5: USB Audio (hw:3,0)(ALSA);2;2; 7;sysdefault(ALSA);128;128; 8;front(ALSA);0;2; 9;surround40(ALSA);0;2; 10;iec958(ALSA);0;2; 11;spdif(ALSA);2;2; 12;pulse(ALSA);32;32;r,i,o 13;upmix(ALSA);8;8; 14;vdownmix(ALSA);6;6; 15;dmix(ALSA);0;2; 16;default(ALSA);32;32; freeswitch at tux> best, Tamer On 2020-02-03 01:43, Tom Lynn wrote: > One more item, I still don't have pulseaudio installed and the pa > commands work, so pulseaudio is not strictly necessary. > > On Sun, Feb 2, 2020 at 4:33 PM Tom Lynn > wrote: > > I followed your steps, and yes, pa commands function when > freeswitch is run from the command line.  Nice! > > I tried running it as both root and as user tom.  Both times I was > able to use the pa commands with success.  I tried running user > tom group user from the systemd startup and was not successful > using pa commands. > > The question becomes, why does this not work using the > freeswitch.service configuration file supplied with the packaged > freeswitch?  I installed freeswitch-meta-all. > > Tom > > On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users > > wrote: > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Sun, 2 Feb 2020 18:02:09 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > Dear Tom, > > This is not true. > I am running freeswitch as the user I am logged in. > > as root: > chown -R tamer:users /opt/freeswitch > > as user: > /opt/freeswitch/bin/freeswitch -nc > > as user: > fs_cli -r > > and you will see portaudio runs without any problems. > > I have it in combination with pulseaudio running. > > > best, Tamer > > On 2020-02-02 17:01, Tom Lynn wrote: > > I do not have pulse audio installed. > > > > I can access alsamixer and aplay from the shell while logged > in as > > either root or freeswitch without issues. > > > > mod_portaudio only works WHEN I'm running freeswitch under > group root, > > which I will not be able to do in a production environment. > > > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via > FreeSWITCH-users > > > > >> wrote: > > > > > > > > > >     ---------- Forwarded message ---------- > >     From: Tamer Higazi > >      >> > >     To: freeswitch-users at lists.freeswitch.org > > >      > > >     Cc: > >     Bcc: > >     Date: Sun, 2 Feb 2020 15:22:31 +0100 > >     Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > >     permissions issue > >     Dear Mark, > > > >     I tell you exactly why. > > > >     That is less a permission issue. > > > >     if you open the shell as "root" and type alsamixer you > won't be > >     capable > >     to access the soundcard because "root" is not in the > 1st) sound group > >     and 2nd) root ha no access to the pulseaudio server > > > >     This is why I MUST run freeswitch as the user that is > currently > >     logged in. > > > >     mod_alsa doesn't work at all. I didn't get it running, only > >     mod_portaudio.... > > > > > >     best, Tamer > > > > > >     On 2020-02-02 07:59, Tom Lynn wrote: > >     > For what it's worth, I'm running into this very same > issue over 4 > >     > years after the original poster.  I suspect that > leaving off the > >     group > >     > option is leaving the process running as group root, > which would > >     mimic > >     > my attempt at having the service run as user > freeswitch/group root, > >     > which allows mod_portaudio to function. > >     > > >     > I've posted here in the mail list, on IRC and in > Slack, but the > >     only > >     > bite was someone hinting about freeswitch user needing > to be in > >     group > >     > audio, which it is.  This is broken. I've looked at > submitting a > >     JIRA > >     > on this, but my login doesn't appear to allow it. > >     > > >     > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > >     > > >     > > >>> wrote: > >     > > >     > covici at ccs.covici.com > > > >      >> > >     >     [covici at ccs.covici.com > > > >      >>] wrote: > >     >     > If  using alsa, did you check the permission of > /dev/snd > >     and its > >     >     > children? > >     > > >     >     Yes.  In fact, both alsa operations (aplay) and > portaudio > >     >     enumeration work > >     >     correctly when running as the freeswitch user.  > All of this is > >     >     detailed in > >     >     my initial post, two or three weeks ago. > >     > > >     >     Mark > >     > > >     > > > >   _________________________________________________________________________ > >     >     Professional FreeSWITCH Consulting Services: > >     > consulting at freeswitch.org > > > > >      > >> > >     > http://www.freeswitchsolutions.com > >     > > >     >     Official FreeSWITCH Sites > >     > http://www.freeswitch.org > >     > http://confluence.freeswitch.org > >     > http://www.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 > >     > http://www.freeswitch.org > >     > > >     > > >     > > > >  _________________________________________________________________________ > >     > > >     > The FreeSWITCH project is sponsored by SignalWire > > https://signalwire.com > >     > Enhance your FreeSWITCH install with disruptive priced > SMS and > >     PSTN services. > >     > Build your next product on our scalable cloud platform. > >     > > >     > Join our online community to chat in real time > > https://signalwire.community > >     > > >     > Professional FreeSWITCH Services > >     > sales at freeswitch.com > > > >     > https://freeswitch.com > >     > > >     > Official FreeSWITCH Sites > >     > https://freeswitch.com/oss > >     > https://freeswitch.org/confluence > >     > https://cluecon.com > >     > > >     > FreeSWITCH-users mailing list > >     > FreeSWITCH-users at lists.freeswitch.org > > >      > > >     > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >     > > >    >  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >     > https://freeswitch.com > > > > > > > > > >     ---------- Forwarded message ---------- > >     From: Tamer Higazi via FreeSWITCH-users > >      > >      >> > >     To: freeswitch-users at lists.freeswitch.org > > >      > > >     Cc: > >     Bcc: > >     Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > >     Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > >     permissions issue > > >  _________________________________________________________________________ > > > >     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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users > > > To: freeswitch-users at lists.freeswitch.org > > Cc: > Bcc: > Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > _________________________________________________________________________ > > 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 tom at tomlynn.com Mon Feb 3 05:46:17 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sun, 2 Feb 2020 21:46:17 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: Thank you, Tamer! I'll try your unit file sometime this week. On Sun, Feb 2, 2020 at 7:58 PM Tamer Higazi via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 3 Feb 2020 04:57:26 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > Dear Tom, > > It doesn't matter. It works, it shows you all devices. > > If you can run as usual user "alsamixer" and be capable to list the > soundcard and you run freeswitch where all files are owned by this user, > and start freeswitch as well as this user then you can access your > soundcard with mod_portaudio without any problems. > > here is my unit file: > > [Unit] > Description=freeswitch > After=syslog.target network.target local-fs.target > > [Service] > User=tamer > Group=users > Type=forking > PIDFile=/opt/freeswitch/var/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/opt/freeswitch/bin/freeswitch -nc > TimeoutSec=45s > Restart=on-failure > ; exec > WorkingDirectory=/opt/freeswitch/bin > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > WantedBy=multi-user.target > > > here: > > tamer at tux ~ $ fs_cli -r > freeswitch at tux> pa dev > > [ devlist] > > > freeswitch at tux> pa devlist > 0;HDA NVidia: HDMI 0 (hw:0,3)(ALSA);0;8; > 1;HDA NVidia: HDMI 1 (hw:0,7)(ALSA);0;8; > 2;HDA NVidia: HDMI 2 (hw:0,8)(ALSA);0;2; > 3;HDA NVidia: HDMI 3 (hw:0,9)(ALSA);0;8; > 4;HD-Audio Generic: ALC892 Analog (hw:1,0)(ALSA);2;2; > 5;Logitech Webcam C925e: USB Audio (hw:2,0)(ALSA);2;0; > 6;Sound Blaster E5: USB Audio (hw:3,0)(ALSA);2;2; > 7;sysdefault(ALSA);128;128; > 8;front(ALSA);0;2; > 9;surround40(ALSA);0;2; > 10;iec958(ALSA);0;2; > 11;spdif(ALSA);2;2; > 12;pulse(ALSA);32;32;r,i,o > 13;upmix(ALSA);8;8; > 14;vdownmix(ALSA);6;6; > 15;dmix(ALSA);0;2; > 16;default(ALSA);32;32; > > freeswitch at tux> > > best, Tamer > > On 2020-02-03 01:43, Tom Lynn wrote: > > One more item, I still don't have pulseaudio installed and the pa > > commands work, so pulseaudio is not strictly necessary. > > > > On Sun, Feb 2, 2020 at 4:33 PM Tom Lynn > > wrote: > > > > I followed your steps, and yes, pa commands function when > > freeswitch is run from the command line. Nice! > > > > I tried running it as both root and as user tom. Both times I was > > able to use the pa commands with success. I tried running user > > tom group user from the systemd startup and was not successful > > using pa commands. > > > > The question becomes, why does this not work using the > > freeswitch.service configuration file supplied with the packaged > > freeswitch? I installed freeswitch-meta-all. > > > > Tom > > > > On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users > > > > wrote: > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi > > > > To: freeswitch-users at lists.freeswitch.org > > > > Cc: > > Bcc: > > Date: Sun, 2 Feb 2020 18:02:09 +0100 > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > Dear Tom, > > > > This is not true. > > I am running freeswitch as the user I am logged in. > > > > as root: > > chown -R tamer:users /opt/freeswitch > > > > as user: > > /opt/freeswitch/bin/freeswitch -nc > > > > as user: > > fs_cli -r > > > > and you will see portaudio runs without any problems. > > > > I have it in combination with pulseaudio running. > > > > > > best, Tamer > > > > On 2020-02-02 17:01, Tom Lynn wrote: > > > I do not have pulse audio installed. > > > > > > I can access alsamixer and aplay from the shell while logged > > in as > > > either root or freeswitch without issues. > > > > > > mod_portaudio only works WHEN I'm running freeswitch under > > group root, > > > which I will not be able to do in a production environment. > > > > > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via > > FreeSWITCH-users > > > > > > > > >> wrote: > > > > > > > > > > > > > > > ---------- Forwarded message ---------- > > > From: Tamer Higazi > > > > > >> > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > > Cc: > > > Bcc: > > > Date: Sun, 2 Feb 2020 15:22:31 +0100 > > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > > permissions issue > > > Dear Mark, > > > > > > I tell you exactly why. > > > > > > That is less a permission issue. > > > > > > if you open the shell as "root" and type alsamixer you > > won't be > > > capable > > > to access the soundcard because "root" is not in the > > 1st) sound group > > > and 2nd) root ha no access to the pulseaudio server > > > > > > This is why I MUST run freeswitch as the user that is > > currently > > > logged in. > > > > > > mod_alsa doesn't work at all. I didn't get it running, only > > > mod_portaudio.... > > > > > > > > > best, Tamer > > > > > > > > > On 2020-02-02 07:59, Tom Lynn wrote: > > > > For what it's worth, I'm running into this very same > > issue over 4 > > > > years after the original poster. I suspect that > > leaving off the > > > group > > > > option is leaving the process running as group root, > > which would > > > mimic > > > > my attempt at having the service run as user > > freeswitch/group root, > > > > which allows mod_portaudio to function. > > > > > > > > I've posted here in the mail list, on IRC and in > > Slack, but the > > > only > > > > bite was someone hinting about freeswitch user needing > > to be in > > > group > > > > audio, which it is. This is broken. I've looked at > > submitting a > > > JIRA > > > > on this, but my login doesn't appear to allow it. > > > > > > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > > > > > > > > > > > > >>> wrote: > > > > > > > > covici at ccs.covici.com > > > > > > > > >> > > > > [covici at ccs.covici.com > > > > > > > > > >>] wrote: > > > > > If using alsa, did you check the permission of > > /dev/snd > > > and its > > > > > children? > > > > > > > > Yes. In fact, both alsa operations (aplay) and > > portaudio > > > > enumeration work > > > > correctly when running as the freeswitch user. > > All of this is > > > > detailed in > > > > my initial post, two or three weeks ago. > > > > > > > > Mark > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > > > > > > > > > > >> > > > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://confluence.freeswitch.org > > > > http://www.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 > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > The FreeSWITCH project is sponsored by SignalWire > > > https://signalwire.com > > > > Enhance your FreeSWITCH install with disruptive priced > > SMS and > > > PSTN services. > > > > Build your next product on our scalable cloud platform. > > > > > > > > Join our online community to chat in real time > > > https://signalwire.community > > > > > > > > Professional FreeSWITCH Services > > > > sales at freeswitch.com > > > > > > > https://freeswitch.com > > > > > > > > Official FreeSWITCH Sites > > > > https://freeswitch.com/oss > > > > https://freeswitch.org/confluence > > > > https://cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > https://freeswitch.com > > > > > > > > > > > > > > > ---------- Forwarded message ---------- > > > From: Tamer Higazi via FreeSWITCH-users > > > > > > > > >> > > > To: freeswitch-users at lists.freeswitch.org > > > > > > > > > > Cc: > > > Bcc: > > > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > > permissions issue > > > > > > _________________________________________________________________________ > > > > > > 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 > > > > > > > > > > ---------- Forwarded message ---------- > > From: Tamer Higazi via FreeSWITCH-users > > > > > > To: freeswitch-users at lists.freeswitch.org > > > > Cc: > > Bcc: > > Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > > permissions issue > > > _________________________________________________________________________ > > > > 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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sun, 02 Feb 2020 19:58:09 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue > _________________________________________________________________________ > > 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 th982a at googlemail.com Mon Feb 3 07:04:27 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Mon, 3 Feb 2020 08:04:27 +0100 Subject: setting mod_portaudio priority codec call out Message-ID: <4a8bbec5-463a-6c69-0aa0-1d62023b1ad0@googlemail.com> Hi people, I am looking to configure the priority codec with mod_portaudio. When I make an outbound call, pa uses PCMA as primary codec... When I receive from the same person a call, G722 is used. Any chance enforcing G722 as priority codec with PCMA as fallback by placing an outbound call? Thank you console output: o=root 1240735378 1240735378 IN IP4 21x.xx.xxx.xxx s=Asterisk PBX 11.21.1 c=IN IP4 21x.xx.xxx.xxx t=0 0 m=audio 12878 RTP/AVP 8 0 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2020-02-03 07:56:26.478728 [NOTICE] sofia.c:7314 Pre-Answer sofia/external/xxxxxxxxxx! 2020-02-03 07:56:26.478728 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.478728 [DEBUG] switch_channel.c:3565 (sofia/external/xxxxxxxxxx) Callstate Change DOWN -> EARLY 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101 at 8000 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/xxxxxxxxxx PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-03 07:56:26.478728 [DEBUG] switch_core_codec.c:111 sofia/external/xxxxxxxxxx Original read codec set to PCMA:8 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101 at 8000 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:5909 sofia/external/xxxxxxxxxx Set 2833 dtmf send payload to 101 recv payload to 101 2020-02-03 07:56:26.478728 [DEBUG] switch_core_media.c:8659 AUDIO RTP [sofia/external/xxxxxxxxxx] 84.75.67.75 port 32108 -> 21x.xx.xxx.xxx port 12878 codec: 8 ms: 20 2020-02-03 07:56:26.478728 [DEBUG] switch_rtp.c:4408 Starting timer [soft] 160 bytes per 20ms 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:8973 sofia/external/xxxxxxxxxx Set 2833 dtmf send payload to 101 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:8980 sofia/external/xxxxxxxxxx Set 2833 dtmf receive payload to 101 2020-02-03 07:56:26.498733 [DEBUG] switch_core_media.c:9003 sofia/external/xxxxxxxxxx Set rtp dtmf delay to 40 2020-02-03 07:56:26.498733 [DEBUG] switch_ivr_originate.c:3809 Originate Resulted in Success: [sofia/external/xxxxxxxxxx] 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] switch_channel.c:2130 (portaudio/1xxxxxxxxxx) Callstate Change RING_WAIT -> ACTIVE 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] mod_portaudio.c:809 portaudio/1xxxxxxxxxx CHANNEL KILL 2020-02-03 07:56:26.498733 [DEBUG] switch_ivr_bridge.c:1796 (sofia/external/xxxxxxxxxx) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2020-02-03 07:56:26.498733 [DEBUG] switch_core_state_machine.c:585 (sofia/external/xxxxxxxxxx) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 2) From andrew at cassidywebservices.co.uk Mon Feb 3 10:57:26 2020 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 3 Feb 2020 10:57:26 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support Message-ID: Hi Guys, I have a client looking for a multi-tenant PBX system with commercial support. I have already put forward Kazoo but they're after a number of options. Any and all suggestions are welcome! Kind regards, -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk www.cassidyweb.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Mon Feb 3 11:10:44 2020 From: joelists at tm.net.uk (Joseph Waite) Date: Mon, 3 Feb 2020 11:10:44 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: References: Message-ID: <29348D4C-D7BD-48D1-885A-CEEB2C9BD92B@tm.net.uk> FusionPBX is the best I have used and I have tried most over the years! Mark Crane and his team offer various commercial support options. Joe Waite > On 3 Feb 2020, at 10:59, Andrew Cassidy wrote: > >  > Hi Guys, > > I have a client looking for a multi-tenant PBX system with commercial support. I have already put forward Kazoo but they're after a number of options. > > Any and all suggestions are welcome! > > Kind regards, > > -- > Andrew Cassidy BSc (Hons) MBCS > Managing Director > > 0330 44 55 960 andrew at cassidyweb.co.uk www.cassidyweb.co.uk > _________________________________________________________________________ > > 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 Mon Feb 3 12:00:35 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 3 Feb 2020 12:00:35 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: <29348D4C-D7BD-48D1-885A-CEEB2C9BD92B@tm.net.uk> References: <29348D4C-D7BD-48D1-885A-CEEB2C9BD92B@tm.net.uk> Message-ID: I don’t use it as much, but it is pretty good. I’d recommend it as well. Unless you want to do some non-standard stuff. On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: > FusionPBX is the best I have used and I have tried most over the years! > > Mark Crane and his team offer various commercial support options. > > Joe Waite > > On 3 Feb 2020, at 10:59, Andrew Cassidy > wrote: > >  > Hi Guys, > > I have a client looking for a multi-tenant PBX system with commercial > support. I have already put forward Kazoo but they're after a number of > options. > > Any and all suggestions are welcome! > > Kind regards, > > -- > *Andrew Cassidy BSc (Hons) MBCS* > Managing Director > > 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk > www.cassidyweb.co.uk > _________________________________________________________________________ > > 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 david.villasmil.work at gmail.com Mon Feb 3 12:01:28 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 3 Feb 2020 12:01:28 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: References: <29348D4C-D7BD-48D1-885A-CEEB2C9BD92B@tm.net.uk> Message-ID: I don’t know how well it scales, though. On Mon, 3 Feb 2020 at 12:00, David Villasmil wrote: > I don’t use it as much, but it is pretty good. I’d recommend it as well. > Unless you want to do some non-standard stuff. > > On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: > >> FusionPBX is the best I have used and I have tried most over the years! >> >> Mark Crane and his team offer various commercial support options. >> >> Joe Waite >> >> On 3 Feb 2020, at 10:59, Andrew Cassidy >> wrote: >> >>  >> Hi Guys, >> >> I have a client looking for a multi-tenant PBX system with commercial >> support. I have already put forward Kazoo but they're after a number of >> options. >> >> Any and all suggestions are welcome! >> >> Kind regards, >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS* >> Managing Director >> >> 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk >> www.cassidyweb.co.uk >> _________________________________________________________________________ >> >> 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 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Mon Feb 3 12:18:20 2020 From: joelists at tm.net.uk (Joseph Waite) Date: Mon, 3 Feb 2020 12:18:20 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: References: Message-ID: It can scale pretty well. There are a number of different options, including clustering etc. Joe Waite > On 3 Feb 2020, at 12:03, David Villasmil wrote: > >  > I don’t know how well it scales, though. > >> On Mon, 3 Feb 2020 at 12:00, David Villasmil wrote: >> I don’t use it as much, but it is pretty good. I’d recommend it as well. Unless you want to do some non-standard stuff. >> >>> On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: >>> FusionPBX is the best I have used and I have tried most over the years! >>> >>> Mark Crane and his team offer various commercial support options. >>> >>> Joe Waite >>> >>>>> On 3 Feb 2020, at 10:59, Andrew Cassidy wrote: >>>>> >>>>  >>>> Hi Guys, >>>> >>>> I have a client looking for a multi-tenant PBX system with commercial support. I have already put forward Kazoo but they're after a number of options. >>>> >>>> Any and all suggestions are welcome! >>>> >>>> Kind regards, >>>> >>>> -- >>>> Andrew Cassidy BSc (Hons) MBCS >>>> Managing Director >>>> >>>> 0330 44 55 960 andrew at cassidyweb.co.uk www.cassidyweb.co.uk >>>> _________________________________________________________________________ >>>> >>>> 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 > -- > 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 lloyd.aloysius at gmail.com Mon Feb 3 14:28:20 2020 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 3 Feb 2020 09:28:20 -0500 Subject: [Freeswitch-users] Debian 9 or 10 for new installs Message-ID: Hi All Looking for a recommendation. Does Debian 10 works reliably for freeswitch v1.10 or stick with Debian 9. Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: From mbodbg at gmx.net Mon Feb 3 15:07:21 2020 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Mon, 3 Feb 2020 16:07:21 +0100 Subject: [Freeswitch-users] Unset variable in dialstring Message-ID: Hello NG, I know it is possible to unset a variable in the dial plan by using the „unset“ application like: References: Message-ID: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Hi, You can try with _undef_, it works in dialplan as Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 3 Feb 2020, at 16:07, Markus Bönke wrote: > > Hello NG, > > I know it is possible to unset a variable in the dial plan by using the „unset“ application like: > > > _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? > > Thanks and regards > > Markus > > > > _________________________________________________________________________ > > 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 brians at iptel.co Mon Feb 3 21:56:59 2020 From: brians at iptel.co (Brian :) Date: Mon, 3 Feb 2020 21:56:59 +0000 Subject: [Freeswitch-users] Debian 9 or 10 for new installs In-Reply-To: References: Message-ID: Debian 10 the recommended distribution for 1.10 On Monday, February 3, 2020, Lloyd Aloysius wrote: > Hi All > Looking for a recommendation. Does Debian 10 works reliably for freeswitch v1.10 or stick with Debian 9. > Thanks > Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakumarsoftware at gmail.com Tue Feb 4 18:16:32 2020 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Tue, 4 Feb 2020 10:16:32 -0800 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: David, Did you make it work ? I'm also running into the same issue. Thanks Shaks On Mon, Dec 16, 2019 at 1:25 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > 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 > _________________________________________________________________________ > > 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 fs at voice2net.ca Tue Feb 4 19:10:22 2020 From: fs at voice2net.ca (fs at voice2net.ca) Date: Tue, 04 Feb 2020 12:10:22 -0700 Subject: [Freeswitch-users] Retrieving Original CLID on Parked Call In-Reply-To: References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: <5688c81c65c6e11115d229082ed1c0f3@voice2net.ca> We are long time freeswitch users. Trying to resolve a request on a park retrieve. Is there a way to display the original Caller ID when you retrieve a parked call and send it to the retrieving device for display. Darcy Primrose Voice2Net From mbodbg at gmx.net Tue Feb 4 19:44:58 2020 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Tue, 4 Feb 2020 20:44:58 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> References: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Message-ID: Unfortunately I doesn’t work. I tried: and see Privacy: _undef_ In the sip INVITE message. Thanks Markus > Am 03.02.2020 um 16:48 schrieb Vallimamod Abdullah : > > Hi, > > You can try with _undef_, it works in dialplan as > > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sip.solutions > linkedin.com/in/vallimamod > . > > >> On 3 Feb 2020, at 16:07, Markus Bönke wrote: >> >> Hello NG, >> >> I know it is possible to unset a variable in the dial plan by using the „unset“ application like: >> >> > >> _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? >> >> Thanks and regards >> >> Markus >> >> >> >> _________________________________________________________________________ >> >> 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 vma at vallimamod.org Tue Feb 4 22:20:02 2020 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Tue, 4 Feb 2020 23:20:02 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: References: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Message-ID: Hi, For sip headers you need to set an empty value to remove them: {sip_h_Privacy=} or before the bridge: the "_nolocal_" (or "nolocal:") prefix tells freeswitch to set the var only on b-leg. Hope this helps. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 4 Feb 2020, at 20:44, Markus Bönke wrote: > > Unfortunately I doesn’t work. I tried: > > > > and see > > Privacy: _undef_ > > In the sip INVITE message. > > Thanks > > Markus > >> Am 03.02.2020 um 16:48 schrieb Vallimamod Abdullah : >> >> Hi, >> >> You can try with _undef_, it works in dialplan as >> >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sip.solutions >> linkedin.com/in/vallimamod >> . >> >> >>> On 3 Feb 2020, at 16:07, Markus Bönke wrote: >>> >>> Hello NG, >>> >>> I know it is possible to unset a variable in the dial plan by using the „unset“ application like: >>> >>> >> >>> _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? >>> >>> Thanks and regards >>> >>> Markus >>> >>> >>> >>> _________________________________________________________________________ >>> >>> 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 mbodbg at gmx.net Wed Feb 5 06:15:35 2020 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Wed, 5 Feb 2020 07:15:35 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: References: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Message-ID: Perfect, that works! Thank you Markus > Am 04.02.2020 um 23:20 schrieb Vallimamod Abdullah : > > Hi, > > For sip headers you need to set an empty value to remove them: {sip_h_Privacy=} > > or before the bridge: > > > > the "_nolocal_" (or "nolocal:") prefix tells freeswitch to set the var only on b-leg. > > Hope this helps. > > > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sip.solutions > linkedin.com/in/vallimamod > . > > >> On 4 Feb 2020, at 20:44, Markus Bönke wrote: >> >> Unfortunately I doesn’t work. I tried: >> >> >> >> and see >> >> Privacy: _undef_ >> >> In the sip INVITE message. >> >> Thanks >> >> Markus >> >>> Am 03.02.2020 um 16:48 schrieb Vallimamod Abdullah : >>> >>> Hi, >>> >>> You can try with _undef_, it works in dialplan as >>> >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sip.solutions >>> linkedin.com/in/vallimamod >>> . >>> >>> >>>> On 3 Feb 2020, at 16:07, Markus Bönke wrote: >>>> >>>> Hello NG, >>>> >>>> I know it is possible to unset a variable in the dial plan by using the „unset“ application like: >>>> >>>> >>> >>>> _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? >>>> >>>> Thanks and regards >>>> >>>> Markus >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 mbodbg at gmx.net Wed Feb 5 09:52:09 2020 From: mbodbg at gmx.net (=?utf-8?Q?Markus_B=C3=B6nke?=) Date: Wed, 5 Feb 2020 10:52:09 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: References: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Message-ID: I made a couple of more tests and found out that it depends on the cid_type. So if the sip_cid_type is set to rpid (or not set as the default is rpid) then the deletion of the Privacy header works: EXECUTE [depth=0] sofia/internal/+4999999999999 at 192.168.12.13 bridge({sip_cid_type=rpid,_nolocal_sip_h_Privacy=}sofia/gateway/asterisk/+4933333333332) INVITE sip:+4933333333332 at 192.168.12.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.2;rport;branch=z9hG4bK55vN1XK3UZNHQ Max-Forwards: 69 From: "+4999999999999" ;tag=Fyr8m981N4Dea To: Call-ID: 0f549364-c29d-1238-6baf-001c42a16581 CSeq: 15893737 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-13-f7bdd3845a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 220 Remote-Party-ID: "+4999999999999" ;party=calling;screen=yes;privacy=off but when the cid_type is set to pid, setting "_nolocal_sip_h_Privacy=" has no effect, the Privacy header is kept. EXECUTE [depth=0] sofia/internal/+4999999999999 at 192.168.12.13 bridge({sip_cid_type=pid,_nolocal_sip_h_Privacy=}sofia/gateway/asterisk/+4933333333332) INVITE sip:+4933333333332 at 192.168.12.18 SIP/2.0 Via: SIP/2.0/UDP 192.168.12.2;rport;branch=z9hG4bK2aHBvc1r44jSm Max-Forwards: 69 From: "+4999999999999" ;tag=Dc6pHK7tUj18j To: Call-ID: c34ec1cd-c29c-1238-6baf-001c42a16581 CSeq: 15893673 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-13-f7bdd3845a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 220 P-Asserted-Identity: "+4999999999999" Is this by design or a bug? Thanks Markus > Am 05.02.2020 um 07:15 schrieb Markus Bönke : > > Perfect, that works! > > Thank you > > Markus >> Am 04.02.2020 um 23:20 schrieb Vallimamod Abdullah : >> >> Hi, >> >> For sip headers you need to set an empty value to remove them: {sip_h_Privacy=} >> >> or before the bridge: >> >> >> >> the "_nolocal_" (or "nolocal:") prefix tells freeswitch to set the var only on b-leg. >> >> Hope this helps. >> >> >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sip.solutions >> linkedin.com/in/vallimamod >> . >> >> >>> On 4 Feb 2020, at 20:44, Markus Bönke wrote: >>> >>> Unfortunately I doesn’t work. I tried: >>> >>> >>> >>> and see >>> >>> Privacy: _undef_ >>> >>> In the sip INVITE message. >>> >>> Thanks >>> >>> Markus >>> >>>> Am 03.02.2020 um 16:48 schrieb Vallimamod Abdullah : >>>> >>>> Hi, >>>> >>>> You can try with _undef_, it works in dialplan as >>>> >>>> >>>> Best Regards, >>>> -- >>>> Vallimamod Abdullah >>>> SIP Solutions >>>> vma at sip.solutions >>>> linkedin.com/in/vallimamod >>>> . >>>> >>>> >>>>> On 3 Feb 2020, at 16:07, Markus Bönke wrote: >>>>> >>>>> Hello NG, >>>>> >>>>> I know it is possible to unset a variable in the dial plan by using the „unset“ application like: >>>>> >>>>> >>>> >>>>> _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? >>>>> >>>>> Thanks and regards >>>>> >>>>> Markus >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 sdevoy at bizfocused.com Wed Feb 5 15:20:39 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 5 Feb 2020 15:20:39 +0000 Subject: [Freeswitch-users] Dropped calls Message-ID: Hi, I have a customer that has reported LOTS of dropped calls over a 6-month period. She claims several a day. I have replaced the phone and the network cabling with no change. I have reviewed many log files. There are no errors reported and the calls all end with a hang-up, mostly from her leg but occasionally the other end. It happens inbound and outbound. It has just occurred to me that if the RTP stopped working, it would appear to be a dropped call to the user who would hang-up. Other users at this site report occasional or rare dropped calls, but the main problem is then receptionist who clearly gets most of the calls. They are NAT connections through a SonicWall firewall that I hate and may be the problem. The server (FS 1.8) is shared with other users/companies and is not NAT. Does anyone have any suggestions for diagnosing this maddening problem? Thank you, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Feb 6 00:10:07 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 6 Feb 2020 01:10:07 +0100 Subject: [Freeswitch-users] Dropped calls In-Reply-To: References: Message-ID: Get the xml cdrs for those dropped calls and check the rtp statistics. Might you you a clue. On Wed, 5 Feb 2020 at 10:25, Sean Devoy wrote: > Hi, > > > > I have a customer that has reported LOTS of dropped calls over a 6-month > period. She claims several a day. I have replaced the phone and the > network cabling with no change. I have reviewed many log files. There are > no errors reported and the calls all end with a hang-up, mostly from her > leg but occasionally the other end. It happens inbound and outbound. It > has just occurred to me that if the RTP stopped working, it would appear to > be a dropped call to the user who would hang-up. Other users at this site > report occasional or rare dropped calls, but the main problem is then > receptionist who clearly gets most of the calls. > > > > They are NAT connections through a SonicWall firewall that I hate and may > be the problem. The server (FS 1.8) is shared with other users/companies > and is not NAT. > > > > Does anyone have any suggestions for diagnosing this maddening problem? > > > > Thank you, > > Sean > _________________________________________________________________________ > > 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 davidswalkabout at gmail.com Thu Feb 6 00:57:41 2020 From: davidswalkabout at gmail.com (David P) Date: Wed, 5 Feb 2020 16:57:41 -0800 Subject: [Freeswitch-users] Verto ice restart Message-ID: I'm looking for best practices about how to detect a) that an ice restart is needed (in verto), and b) how to invoke the restart. FWIW, we use FS v1.8 and will soon try 1.10.2. For detection, I think it could be done either of these ways: a1) In the 1st arg to jQuery.verto(), provide an object with property deviceParams.onICEComplete set to a callback. In the callback, iterate over mediaData.candidateList. For example, check if all candidates are private IPs but all your callers are expected to have public IPs. a2) In the 2nd arg to jQuery.verto(), provide an object with property 'onDialogState' set to a callback. In the callback, if arg1.state.name is "hangup" and arg1.cause is "INCOMPATIBLE DESTINATION". (Ignoring that this hangup cause occurs for other reasons also.) To do the actual restart... In method (a1) we should have access to the peer connection object via arg1.peer. Would it mess up the state of verto to do arg1.peer.createOffer({iceRestart:true}) ? Do we want to limit this reaction to happen just once per call attempt in order to avoid possibly many chained failures? In method (a2), I'm not sure there's any way to access the peer connection. I searched jquery.FSRTC.js for "restart" in case there's a standard way of doing this, but found no match. Cheers, David -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Thu Feb 6 10:05:26 2020 From: mickael at winlux.fr (Mickael Hubert) Date: Thu, 6 Feb 2020 11:05:26 +0100 Subject: [Freeswitch-users] Sip Session Timer on the fly Message-ID: Hi all, I would like to know, if there is a way to FS is "transparent" with SST ? Currently, you can activate or deactivate SST on profile: But I want if client use SST, FS sends it to the B leg, and if client doesn't use SST, FS doesn't send it. thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Thu Feb 6 11:50:19 2020 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 6 Feb 2020 12:50:19 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: References: <1F973D95-3417-42E2-9A91-991E8A155E7C@vallimamod.org> Message-ID: <259E1122-BC08-4CA1-B583-606325069A95@vallimamod.org> Hi, If you want to handle Privacy and P-Asserted-Id headers manually, it is best to set sip_cid_type to "none". Also, if you use the "{...}sofia/gateway/..." syntax, you don't need to add the "_nolocal_" prefix as the vars between the curly braces are only set for the b-leg. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 5 Feb 2020, at 10:52, Markus Bönke wrote: > > I made a couple of more tests and found out that it depends on the cid_type. > So if the sip_cid_type is set to rpid (or not set as the default is rpid) > then the deletion of the Privacy header works: > > > EXECUTE [depth=0] sofia/internal/+4999999999999 at 192.168.12.13 bridge({sip_cid_type=rpid,_nolocal_sip_h_Privacy=}sofia/gateway/asterisk/+4933333333332) > > INVITE sip:+4933333333332 at 192.168.12.18 SIP/2.0 > Via: SIP/2.0/UDP 192.168.12.2;rport;branch=z9hG4bK55vN1XK3UZNHQ > Max-Forwards: 69 > From: "+4999999999999" ;tag=Fyr8m981N4Dea > To: > Call-ID: 0f549364-c29d-1238-6baf-001c42a16581 > CSeq: 15893737 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-13-f7bdd3845a~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 220 > Remote-Party-ID: "+4999999999999" ;party=calling;screen=yes;privacy=off > > > but when the cid_type is set to pid, setting "_nolocal_sip_h_Privacy=" has no effect, the Privacy header is kept. > > EXECUTE [depth=0] sofia/internal/+4999999999999 at 192.168.12.13 bridge({sip_cid_type=pid,_nolocal_sip_h_Privacy=}sofia/gateway/asterisk/+4933333333332) > > > INVITE sip:+4933333333332 at 192.168.12.18 SIP/2.0 > Via: SIP/2.0/UDP 192.168.12.2;rport;branch=z9hG4bK2aHBvc1r44jSm > Max-Forwards: 69 > From: "+4999999999999" ;tag=Dc6pHK7tUj18j > To: > Call-ID: c34ec1cd-c29c-1238-6baf-001c42a16581 > CSeq: 15893673 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.10.2-release-13-f7bdd3845a~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Privacy: none > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 220 > P-Asserted-Identity: "+4999999999999" > > > Is this by design or a bug? > > Thanks > > Markus > >> Am 05.02.2020 um 07:15 schrieb Markus Bönke : >> >> Perfect, that works! >> >> Thank you >> >> Markus >>> Am 04.02.2020 um 23:20 schrieb Vallimamod Abdullah : >>> >>> Hi, >>> >>> For sip headers you need to set an empty value to remove them: {sip_h_Privacy=} >>> >>> or before the bridge: >>> >>> >>> >>> the "_nolocal_" (or "nolocal:") prefix tells freeswitch to set the var only on b-leg. >>> >>> Hope this helps. >>> >>> >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sip.solutions >>> linkedin.com/in/vallimamod >>> . >>> >>> >>>> On 4 Feb 2020, at 20:44, Markus Bönke wrote: >>>> >>>> Unfortunately I doesn’t work. I tried: >>>> >>>> >>>> >>>> and see >>>> >>>> Privacy: _undef_ >>>> >>>> In the sip INVITE message. >>>> >>>> Thanks >>>> >>>> Markus >>>> >>>>> Am 03.02.2020 um 16:48 schrieb Vallimamod Abdullah : >>>>> >>>>> Hi, >>>>> >>>>> You can try with _undef_, it works in dialplan as >>>>> >>>>> >>>>> Best Regards, >>>>> -- >>>>> Vallimamod Abdullah >>>>> SIP Solutions >>>>> vma at sip.solutions >>>>> linkedin.com/in/vallimamod >>>>> . >>>>> >>>>> >>>>>> On 3 Feb 2020, at 16:07, Markus Bönke wrote: >>>>>> >>>>>> Hello NG, >>>>>> >>>>>> I know it is possible to unset a variable in the dial plan by using the „unset“ application like: >>>>>> >>>>>> >>>>> >>>>>> _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? >>>>>> >>>>>> Thanks and regards >>>>>> >>>>>> Markus >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> 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 > > > _________________________________________________________________________ > > 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 Thu Feb 6 17:32:50 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 6 Feb 2020 17:32:50 +0000 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: No, i never got it to work. Freeswitch kept complaining with *Cannot Blind Transfer 1 Legged calls *even when I was answering on FS and executing an echo just for testing. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Feb 4, 2020 at 1:50 PM Sharath Kumar wrote: > David, > Did you make it work ? I'm also running into the same issue. > Thanks > Shaks > > On Mon, Dec 16, 2019 at 1:25 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> 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 >> _________________________________________________________________________ >> >> 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 Thu Feb 6 18:30:01 2020 From: davidswalkabout at gmail.com (David P) Date: Thu, 6 Feb 2020 10:30:01 -0800 Subject: [Freeswitch-users] Can execute_on_media_timeout try another peer? Message-ID: If we lose RTP from a peer and a MEDIA_TIMEOUT occurs on leg B, can execute_on_media_timeout be used to dial that peer again or another one? The "transfer" example at https://freeswitch.org/confluence/display/FREESWITCH/execute_on_media_timeout implies that this is possible, but the example isn't self-explanatory. I didn't find a confluence entry about "transfer" itself but I did find: https://freeswitch.org/confluence/display/FREESWITCH/transfer_on_fail https://freeswitch.org/confluence/display/FREESWITCH/transfer_to Which should be used for my scenario? We'll want to play an audio recording before doing the retry, too. Should we use https://freeswitch.org/confluence/display/FREESWITCH/transfer_ringback Cheers, David -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Thu Feb 6 18:47:29 2020 From: davidswalkabout at gmail.com (David P) Date: Thu, 6 Feb 2020 10:47:29 -0800 Subject: [Freeswitch-users] Can execute_on_media_timeout try another peer? In-Reply-To: References: Message-ID: In particular, we connect to the peer via “conference_set_auto_outcall”. -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdevoy at bizfocused.com Thu Feb 6 19:30:23 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 6 Feb 2020 19:30:23 +0000 Subject: [Freeswitch-users] Dropped calls In-Reply-To: References: Message-ID: Hi David, Thank you very much for the response. I am afraid I must ask you to break that down as it is “beyond my pay grade.” Sorry for being such a novice. “Get the xml cdrs for those dropped calls” – Where do I get these? From the freeswith.log? “check the rtp statistics” – Where? How? Thanks, Sean From: FreeSWITCH-users On Behalf Of David Villasmil Sent: Wednesday, February 5, 2020 7:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Dropped calls Get the xml cdrs for those dropped calls and check the rtp statistics. Might you you a clue. On Wed, 5 Feb 2020 at 10:25, Sean Devoy > wrote: Hi, I have a customer that has reported LOTS of dropped calls over a 6-month period. She claims several a day. I have replaced the phone and the network cabling with no change. I have reviewed many log files. There are no errors reported and the calls all end with a hang-up, mostly from her leg but occasionally the other end. It happens inbound and outbound. It has just occurred to me that if the RTP stopped working, it would appear to be a dropped call to the user who would hang-up. Other users at this site report occasional or rare dropped calls, but the main problem is then receptionist who clearly gets most of the calls. They are NAT connections through a SonicWall firewall that I hate and may be the problem. The server (FS 1.8) is shared with other users/companies and is not NAT. Does anyone have any suggestions for diagnosing this maddening problem? Thank you, Sean _________________________________________________________________________ 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 david.villasmil.work at gmail.com Thu Feb 6 20:39:07 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 6 Feb 2020 15:39:07 -0500 Subject: [Freeswitch-users] Dropped calls In-Reply-To: References: Message-ID: No problem Sean, Did you enable mod_xml_cdr? If so, you should have an xml file for every call that goes through FS in /var/log/freeswitch/xml_cdrs In that file you will find several rtp statistics. You can enable it by un commenting it from autoload_configs/modules.xml Hope that helps. David On Thu, 6 Feb 2020 at 14:56, Sean Devoy wrote: > Hi David, > > Thank you very much for the response. I am afraid I must ask you to break > that down as it is “beyond my pay grade.” Sorry for being such a novice. > > “Get the xml cdrs for those dropped calls” – Where do I get these? From > the freeswith.log? > > “check the rtp statistics” – Where? How? > > > > Thanks, > > Sean > > > > *From:* FreeSWITCH-users *On > Behalf Of *David Villasmil > *Sent:* Wednesday, February 5, 2020 7:10 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Dropped calls > > > > Get the xml cdrs for those dropped calls and check the rtp statistics. > Might you you a clue. > > > > On Wed, 5 Feb 2020 at 10:25, Sean Devoy wrote: > > Hi, > > > > I have a customer that has reported LOTS of dropped calls over a 6-month > period. She claims several a day. I have replaced the phone and the > network cabling with no change. I have reviewed many log files. There are > no errors reported and the calls all end with a hang-up, mostly from her > leg but occasionally the other end. It happens inbound and outbound. It > has just occurred to me that if the RTP stopped working, it would appear to > be a dropped call to the user who would hang-up. Other users at this site > report occasional or rare dropped calls, but the main problem is then > receptionist who clearly gets most of the calls. > > > > They are NAT connections through a SonicWall firewall that I hate and may > be the problem. The server (FS 1.8) is shared with other users/companies > and is not NAT. > > > > Does anyone have any suggestions for diagnosing this maddening problem? > > > > Thank you, > > Sean > > _________________________________________________________________________ > > 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From shakumarsoftware at gmail.com Fri Feb 7 15:29:53 2020 From: shakumarsoftware at gmail.com (Sharath Kumar) Date: Fri, 7 Feb 2020 07:29:53 -0800 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: Thanks, Yeah I have the same problem. Perhaps we can loopback the 2nd leg. I will try and let you know. On Thu, Feb 6, 2020 at 9:54 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > No, i never got it to work. Freeswitch kept complaining with *Cannot > Blind Transfer 1 Legged calls *even when I was answering on FS and > executing an echo just for testing. > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Tue, Feb 4, 2020 at 1:50 PM Sharath Kumar > wrote: > >> David, >> Did you make it work ? I'm also running into the same issue. >> Thanks >> Shaks >> >> On Mon, Dec 16, 2019 at 1:25 PM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> 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 >>> _________________________________________________________________________ >>> >>> 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 Fri Feb 7 16:36:25 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 7 Feb 2020 11:36:25 -0500 Subject: [Freeswitch-users] Refer and Cannot Blind Transfer 1 Legged calls In-Reply-To: References: <1D10EB1D-5A28-4CAA-AEBB-25E89FAE8C37@freeswitch.org> Message-ID: That’ll be great, thanks! On Fri, 7 Feb 2020 at 11:12, Sharath Kumar wrote: > Thanks, Yeah I have the same problem. Perhaps we can loopback the 2nd leg. > I will try and let you know. > > On Thu, Feb 6, 2020 at 9:54 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> No, i never got it to work. Freeswitch kept complaining with *Cannot >> Blind Transfer 1 Legged calls *even when I was answering on FS and >> executing an echo just for testing. >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> >> On Tue, Feb 4, 2020 at 1:50 PM Sharath Kumar >> wrote: >> >>> David, >>> Did you make it work ? I'm also running into the same issue. >>> Thanks >>> Shaks >>> >>> On Mon, Dec 16, 2019 at 1:25 PM David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> 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 >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: 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URL: From vineet.verma at bics.com Mon Feb 10 23:05:01 2020 From: vineet.verma at bics.com (VERMA Vineet (BCS/PSD)) Date: Mon, 10 Feb 2020 23:05:01 +0000 Subject: FreeSWITCH-users Digest, Vol 163, Issue 12 In-Reply-To: References: Message-ID: Dears, I am facing issue on freeswitch release 1.10 that after reaching to 33 calls per second rate. Freeswitch chokes the caps with rate of 16 calls per seconds only. Can you please help me to overcome this issue? Thanks, Vineet verma -----Original Message----- From: FreeSWITCH-users On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: 06 January 2020 18:58 To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 163, Issue 12 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." This e-mail cannot be used for other purposes than BICS business use. See more on https://bics.com/mail-disclaimer From dragos at freeswitch.org Tue Feb 11 15:02:29 2020 From: dragos at freeswitch.org (Dragos Oancea) Date: Tue, 11 Feb 2020 15:02:29 +0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 163, Issue 12 In-Reply-To: References: Message-ID: maybe increase this in switch.conf.xml : On Mon, Feb 10, 2020 at 11:05 PM VERMA Vineet (BCS/PSD) via FreeSWITCH-users wrote: > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/PSD)" > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 10 Feb 2020 23:05:01 +0000 > Subject: RE: FreeSWITCH-users Digest, Vol 163, Issue 12 > Dears, > > I am facing issue on freeswitch release 1.10 that after reaching to 33 > calls per second rate. > Freeswitch chokes the caps with rate of 16 calls per seconds only. > > Can you please help me to overcome this issue? > > Thanks, > Vineet verma > > > -----Original Message----- > From: FreeSWITCH-users On > Behalf Of freeswitch-users-request at lists.freeswitch.org > Sent: 06 January 2020 18:58 > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 163, Issue 12 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific than > "Re: Contents of FreeSWITCH-users digest..." > This e-mail cannot be used for other purposes than BICS business use. See > more on https://bics.com/mail-disclaimer > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/PSD) via FreeSWITCH-users" < > freeswitch-users at lists.freeswitch.org> > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 10 Feb 2020 15:05:43 -0800 (PST) > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 163, Issue 12 > _________________________________________________________________________ > > 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 sdevoy at bizfocused.com Tue Feb 11 18:30:16 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 11 Feb 2020 18:30:16 +0000 Subject: [Freeswitch-users] MultiTenant Message-ID: Hi, I am setting up new server with Multi-Tenant. I have one working already v1.8 on external IP address, but cannot get the new version 1.10 working on a LAN 10.0.0.n. I have gone over these steps many many times: https://freeswitch.org/confluence/display/FREESWITCH/Multi-tenant. I am just trying to get internal extensions working at this point with a FQDN. On the v1.10 server I get registration errors that say: 2020-02-11 13:16:32.986364 [WARNING] sofia_reg.c:2929 Can't find user [111 at 10.0.0.34] from 10.0.0.182 You must define a domain called '10.0.0.34' in your directory and add a user with the id="111" attribute and you must configure your device to use the proper domain in its authentication credentials. However, the phones are registering with the FQDN, not IP address. What am I missing? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Feb 12 08:14:21 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 12 Feb 2020 09:14:21 +0100 Subject: [Freeswitch-users] MultiTenant In-Reply-To: References: Message-ID: Get a sip trace with sngrep or tcpdump ad see what actually happens Or, from fs_cli, "sofia global siptrace on" and watch On Tue, Feb 11, 2020, 20:03 Sean Devoy wrote: > Hi, > > > > I am setting up new server with Multi-Tenant. I have one working already > v1.8 on external IP address, but cannot get the new version 1.10 working on > a LAN 10.0.0.n. I have gone over these steps many many times: > https://freeswitch.org/confluence/display/FREESWITCH/Multi-tenant. I am > just trying to get internal extensions working at this point with a FQDN. > > > > On the v1.10 server I get registration errors that say: > > 2020-02-11 13:16:32.986364 [WARNING] sofia_reg.c:2929 Can't find user [ > 111 at 10.0.0.34] from 10.0.0.182 > > You must define a domain called '10.0.0.34' in your directory and add a > user with the id="111" attribute > > and you must configure your device to use the proper domain in its > authentication credentials. > > > > However, the phones are registering with the FQDN, not IP address. > > > > What am I missing? > > > > Thanks, > > Sean > _________________________________________________________________________ > > 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 arjun.nainwal at startelelogic.co.in Wed Feb 12 12:40:19 2020 From: arjun.nainwal at startelelogic.co.in (Arjunstl) Date: Wed, 12 Feb 2020 05:40:19 -0700 (MST) Subject: [Freeswitch-users] freeswitch Unimrcp Issue Message-ID: <1581511219478-0.post@n2.nabble.com> i want to test freeswitch ASR with unimrcp. in which i want to test only my words of grammar, for an example if i speak certain words and if those words are defined into my grammar then it will go ahead else it will not. can you please help me out regarding this requrement? my environment is : uniMRCP Server : Ubuntu 16.04.6 LTS FreeSWITCH Server : Debian GNU/Linux 9.10 (Stretch) my point of concerns are: 1. i'm testing a dialplan in which i'm loading grammar path from the same server of freeswitch. but its not loading. sending grammar-loading-failure. how to use grammar for scripts. i'm stuck, what and how to load the grammar path for testing. is anything i'm missing dering configuration. 2. i want, when i make a test, then it should recognize only those words which i defined into my grammar or phrases. apart from those words, it shouldn't recognize. can you please help me out to resolve these issues. any help or suggestion, i will be greetful. thanks -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From aingarant at gmail.com Sat Feb 1 05:18:44 2020 From: aingarant at gmail.com (Aingaran Thirunadarajah) Date: Sat, 1 Feb 2020 00:18:44 -0500 Subject: [Freeswitch-users] MOD_SMS and ChatPlan - sofia_presence.c:225 Can't find registered user Message-ID: I am trying to send text between 2 registered extensions using Acrobits Groundwire. Extensions 1000 and 2000. Calls work in & out between the 2 extensions. I created a default.xml chat plan with the following, and reply is successfully sent to the sender. I am trying to forward, redirect the message from the sender to recipient. >From what I understand you can only use the SEND application once, so what is the correct application I must use to redirect the message from 1000 to 2000 or visa versa? Here is the log: (1001 is the sender). 2020-01-31 21:15:31.585237 [INFO] mod_sms.c:368 Processing text message 1001->1002 in context public Chatplan: 1002 parsing [public->demo] continue=false Chatplan: 1002 at 192.168.2.100 Regex (PASS) [demo] to(1002 at 192.168.2.100) =~ /^(.*)$/ break=on-false Chatplan: 1002 at 192.168.2.100 Action info() Chatplan: 1002 at 192.168.2.100 Action reply(BODY: ${_body}. FROM: ${from_user} TO: ${to_user}) Chatplan: 1002 at 192.168.2.100 Action set(to=${to_user}@${to_host}) Chatplan: 1002 at 192.168.2.100 Action set(dest_proto=sip) Chatplan: 1002 at 192.168.2.100 Action set(skip_global_process=true) Chatplan: 1002 at 192.168.2.100 Action set(from=${from_user}) Chatplan: 1002 at 192.168.2.100 Action set(final_delivery=true) Chatplan: 1002 at 192.168.2.100 Action set(from_full=) 2020-01-31 21:15:31.605242 [INFO] mod_sms.c:495 CHANNEL_DATA: Event-Name: [MESSAGE] Core-UUID: [87adf9e7-05fa-4a0e-979c-3a95c8a5dcfb] FreeSWITCH-Hostname: [localhost] FreeSWITCH-Switchname: [localhost] FreeSWITCH-IPv4: [198.98.50.94] FreeSWITCH-IPv6: [::1] Event-Date-Local: [2020-01-31 21:15:31] Event-Date-GMT: [Sat, 01 Feb 2020 05:15:31 GMT] Event-Date-Timestamp: [1580534131585237] Event-Calling-File: [sofia_presence.c] Event-Calling-Function: [sofia_presence_handle_sip_i_message] Event-Calling-Line-Number: [4884] Event-Sequence: [14024] login: [sip:mod_sofia at 198.98.50.94:8700] proto: [sip] to_proto: [sip] from: [1001 at 192.168.2.100] from_user: [1001] from_host: [192.168.2.100] to_user: [1002] to_host: [192.168.2.100] from_sip_ip: [99.244.253.120] from_sip_port: [2833] to: [1002 at 192.168.2.100] subject: [SIMPLE MESSAGE] context: [public] type: [text/plain] from_full: [;tag=AABB1326546BE30A8DB5A5B4E1AA398A] sip_profile: [internal] dest_proto: [sip] max_forwards: [70] DP_MATCH: [1002 at 192.168.2.100] Content-Length: 4 Hi. 2020-01-31 21:15:31.605242 [DEBUG] sofia_presence.c:225 Can't find registered user 1001 at 192.168.2.100 Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From vijaykumar.vadde at gmail.com Sun Feb 2 04:37:28 2020 From: vijaykumar.vadde at gmail.com (Vijaykumar Vadde) Date: Sat, 1 Feb 2020 21:37:28 -0700 Subject: [Freeswitch-users] Restricting In bound calls In-Reply-To: References: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> Message-ID: one option is using parameter "network_addr" . check this link " https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan" On Sat, Feb 1, 2020 at 9:21 PM Tom Lynn wrote: > Have you looked in acl.conf.xml located in autoload_configs? > > On Sat, Feb 1, 2020 at 3:03 PM William Colls > wrote: > >> >> I am sure that I read somewhere that there is a parameter that i can set >> so that only incoming calls from specific IP address are accepted, but I >> can't find the reference, remember the parameter name, or where it is >> located. Could some kind sole point me in the right direction? >> >> Thanks for you time. >> >> William. >> >> >> _________________________________________________________________________ >> >> 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 ochere at gmail.com Sun Feb 2 05:59:30 2020 From: ochere at gmail.com (Frank Ochere) Date: Sun, 2 Feb 2020 08:59:30 +0300 Subject: [Freeswitch-users] Restricting In bound calls In-Reply-To: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> References: <060dc0aa-ac3e-85c7-7753-c569707462bf@williamcollsassoc.ca> Message-ID: Hi William edit external.xml under the sip_profiles directory,under the settings tree add save the file. (If using different file other than external.xml then edit that instead, eg external2.xml, carrier1.xml etc) edit acl.conf.xml and add save the file. on freeswitch console (fs_cli -l4) reloadxml reloadacl you're golden Regards Frank On Sun, Feb 2, 2020 at 2:36 AM William Colls wrote: > > I am sure that I read somewhere that there is a parameter that i can set > so that only incoming calls from specific IP address are accepted, but I > can't find the reference, remember the parameter name, or where it is > located. Could some kind sole point me in the right direction? > > Thanks for you time. > > William. > > > _________________________________________________________________________ > > 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 abalashov at evaristesys.com Mon Feb 3 13:09:06 2020 From: abalashov at evaristesys.com (Alex Balashov) Date: Mon, 3 Feb 2020 08:09:06 -0500 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: References: Message-ID: <351AB48F-BCB4-4176-BA39-FC050E5FCFAE@evaristesys.com> I second the recommendation for Fusion. — Sent from mobile, with due apologies for brevity and errors. > On Feb 3, 2020, at 8:02 AM, David Villasmil wrote: > >  > I don’t know how well it scales, though. > >> On Mon, 3 Feb 2020 at 12:00, David Villasmil wrote: >> I don’t use it as much, but it is pretty good. I’d recommend it as well. Unless you want to do some non-standard stuff. >> >>> On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: >>> FusionPBX is the best I have used and I have tried most over the years! >>> >>> Mark Crane and his team offer various commercial support options. >>> >>> Joe Waite >>> >>>>> On 3 Feb 2020, at 10:59, Andrew Cassidy wrote: >>>>> >>>>  >>>> Hi Guys, >>>> >>>> I have a client looking for a multi-tenant PBX system with commercial support. I have already put forward Kazoo but they're after a number of options. >>>> >>>> Any and all suggestions are welcome! >>>> >>>> Kind regards, >>>> >>>> -- >>>> Andrew Cassidy BSc (Hons) MBCS >>>> Managing Director >>>> >>>> 0330 44 55 960 andrew at cassidyweb.co.uk www.cassidyweb.co.uk >>>> _________________________________________________________________________ >>>> >>>> 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 > -- > 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 Mon Feb 3 14:40:48 2020 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Mon, 3 Feb 2020 15:40:48 +0100 Subject: [Freeswitch-users] Debian 9 or 10 for new installs In-Reply-To: References: Message-ID: Hi Lloyd, It might depend on your needs, I recently ran into issues with FreeTDM when compiling FreeSWITCH on Debian 10, and from what I read with older versions I wouldn't have had the same issue (though in the end I managed on 10). If the vanilla installation works for you Debian 10 is probably the way to go, for me that was a very smooth installation at least. Best, Sam On Mon, Feb 3, 2020 at 3:30 PM Lloyd Aloysius wrote: > Hi All > > Looking for a recommendation. Does Debian 10 works reliably for freeswitch > v1.10 or stick with Debian 9. > > Thanks > Lloyd > _________________________________________________________________________ > > 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 heedfeld at gmail.com Mon Feb 3 15:40:33 2020 From: heedfeld at gmail.com (Henning Heedfeld) Date: Mon, 3 Feb 2020 16:40:33 +0100 Subject: [Freeswitch-users] Unset variable in dialstring In-Reply-To: References: Message-ID: Using _undef_ should work. > Am 03.02.2020 um 16:07 schrieb Markus Bönke : > > Hello NG, > > I know it is possible to unset a variable in the dial plan by using the „unset“ application like: > > > _unset_ does not work with FS 1.10.2 but maybe there is any other „special value“ to unset? > > Thanks and regards > > Markus > > > > _________________________________________________________________________ > > 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 Tue Feb 4 11:48:15 2020 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Tue, 4 Feb 2020 12:48:15 +0100 Subject: [Freeswitch-users] FreeSWITCH and libsng_isdn Message-ID: Hi, I'm wondering if anyone has recent experience using FreeSWITCH 1.10 on Debian 10 with libsng_isdn. I'm compiling from source and getting the errors below. I installed the current libsng_isdn, which is version 8.3.4. However, the FreeTDM instructions mention I need 1.2.0+ which makes me wonder whether these instructions might be outdated. Would an alternative be to just install DAHDI + libpri? I haven't tried that yet but would be happy to. This is all kind of new to me and I'm not entirely sure which components perform which tasks. I have wanpipe installed and am using a PRI/E1 line, I can see that my first port is connected (service wanrouter status), the card is a Sangoma A104. If I uninstall libsng_isdn I'm able to build and install FreeSWITCH with mod_freetdm but the errors below seem to be blocking me when I try to set it up with an ISDN library. Kind regards, Sam CC ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo In file included from /usr/include/sng_isdn/sng_isdn.h:32, from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.h:55, from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.c:38: /usr/include/sng_isdn/ssi.x:1253:20: error: inline function ‘SDisInt’ declared but never defined [-Werror] EXTERN INLINE S16 SDisInt ARGS((void )); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1252:20: error: inline function ‘SEnbInt’ declared but never defined [-Werror] EXTERN INLINE S16 SEnbInt ARGS((void )); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1245:19: error: inline function ‘SSetIntPend’ declared but never defined [-Werror] EXTERN INLINE S16 SSetIntPend ARGS((U16 id, Bool flag)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1244:19: error: inline function ‘SChkNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SChkNxtDBuf ARGS((Buffer *mBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1243:19: error: inline function ‘SGetNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SGetNxtDBuf ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1242:19: error: inline function ‘SInitNxtDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SInitNxtDBuf ARGS((Buffer *mBuf)); ^~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1237:19: error: inline function ‘SUpdMsg’ declared but never defined [-Werror] EXTERN INLINE S16 SUpdMsg ARGS((Buffer *mBuf, Buffer *dBuf, MsgLen mLen)); ^~~~~~~ /usr/include/sng_isdn/ssi.x:1236:19: error: inline function ‘SGetDataTx’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDataTx ARGS((Buffer *dBuf, Data **dat, MsgLen *mLen)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1235:19: error: inline function ‘SGetDataRx’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDataRx ARGS((Buffer *dBuf, MsgLen pad, Data **dat, MsgLen *mLen)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1234:19: error: inline function ‘SRemDBufPre’ declared but never defined [-Werror] EXTERN INLINE S16 SRemDBufPre ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1233:19: error: inline function ‘SRemDBufPst’ declared but never defined [-Werror] EXTERN INLINE S16 SRemDBufPst ARGS((Buffer *mBuf, Buffer **dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1232:19: error: inline function ‘SAddDBufPre’ declared but never defined [-Werror] EXTERN INLINE S16 SAddDBufPre ARGS((Buffer *mBuf, Buffer *dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1231:19: error: inline function ‘SAddDBufPst’ declared but never defined [-Werror] EXTERN INLINE S16 SAddDBufPst ARGS((Buffer *mBuf, Buffer *dBuf)); ^~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1068:20: error: inline function ‘SPutDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SPutDBuf ARGS((Region region,Pool pool,Buffer *buf)); ^~~~~~~~ /usr/include/sng_isdn/ssi.x:1067:20: error: inline function ‘SGetDBuf’ declared but never defined [-Werror] EXTERN INLINE S16 SGetDBuf ARGS((Region region,Pool pool,Buffer * *bufPtr)); ^~~~~~~~ /usr/include/sng_isdn/ssi.x:1058:20: error: inline function ‘SDequeueLast’ declared but never defined [-Werror] EXTERN INLINE S16 SDequeueLast ARGS((Buffer * *bufPtr,Queue *q)); ^~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1057:20: error: inline function ‘SDequeueFirst’ declared but never defined [-Werror] EXTERN INLINE S16 SDequeueFirst ARGS((Buffer * *bufPtr,Queue *q)); ^~~~~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1056:20: error: inline function ‘SQueueLast’ declared but never defined [-Werror] EXTERN INLINE S16 SQueueLast ARGS((Buffer *buf,Queue *q)); ^~~~~~~~~~ /usr/include/sng_isdn/ssi.x:1055:20: error: inline function ‘SQueueFirst’ declared but never defined [-Werror] EXTERN INLINE S16 SQueueFirst ARGS((Buffer *buf,Queue *q)); ^~~~~~~~~~~ cc1: all warnings being treated as errors -------------- next part -------------- An HTML attachment was scrubbed... URL: From svanherwaarden at precisionag.org Tue Feb 4 13:02:20 2020 From: svanherwaarden at precisionag.org (Sam van Herwaarden) Date: Tue, 4 Feb 2020 14:02:20 +0100 Subject: [Freeswitch-users] FreeSWITCH and libsng_isdn In-Reply-To: References: Message-ID: Hi, It seems I figured it out. The "inline function X declared but never defined" is actually a warning, not an error, but I think Debian 10 switched to a newer version of GCC which is stricter with warnings (?). If I use CFLAGS="... -Wno-error" MOD_CFLAGS="... -Wno-error" ./configure instead of just ./configure the compilation doesn't halt on this issue. Next I did have another issue, which is that libsng_isdn gets installed to /usr/lib64 but FreeSWITCH can't find it there (I guess this is an ld configuration issue). A mv /usr/lib64/libsng* /usr/lib/ took care of that. Kind regards, Sam On Tue, Feb 4, 2020 at 12:48 PM Sam van Herwaarden < svanherwaarden at precisionag.org> wrote: > Hi, > > I'm wondering if anyone has recent experience using FreeSWITCH 1.10 on > Debian 10 with libsng_isdn. I'm compiling from source and getting the > errors below. > > I installed the current libsng_isdn, which is version 8.3.4. However, the FreeTDM > instructions > mention I > need 1.2.0+ which makes me wonder whether these instructions might be > outdated. > > Would an alternative be to just install DAHDI + libpri? I haven't tried > that yet but would be happy to. This is all kind of new to me and I'm not > entirely sure which components perform which tasks. > > I have wanpipe installed and am using a PRI/E1 line, I can see that my > first port is connected (service wanrouter status), the card is a Sangoma > A104. If I uninstall libsng_isdn I'm able to build and install FreeSWITCH > with mod_freetdm but the errors below seem to be blocking me when I try to > set it up with an ISDN library. > > Kind regards, > Sam > > CC ftmod_sangoma_isdn_la-ftmod_sangoma_isdn.lo > In file included from /usr/include/sng_isdn/sng_isdn.h:32, > from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.h:55, > from src/ftmod/ftmod_sangoma_isdn/ftmod_sangoma_isdn.c:38: > /usr/include/sng_isdn/ssi.x:1253:20: error: inline function ‘SDisInt’ > declared but never defined [-Werror] > EXTERN INLINE S16 SDisInt ARGS((void )); > ^~~~~~~ > /usr/include/sng_isdn/ssi.x:1252:20: error: inline function ‘SEnbInt’ > declared but never defined [-Werror] > EXTERN INLINE S16 SEnbInt ARGS((void )); > ^~~~~~~ > /usr/include/sng_isdn/ssi.x:1245:19: error: inline function ‘SSetIntPend’ > declared but never defined [-Werror] > EXTERN INLINE S16 SSetIntPend ARGS((U16 id, Bool flag)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1244:19: error: inline function ‘SChkNxtDBuf’ > declared but never defined [-Werror] > EXTERN INLINE S16 SChkNxtDBuf ARGS((Buffer *mBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1243:19: error: inline function ‘SGetNxtDBuf’ > declared but never defined [-Werror] > EXTERN INLINE S16 SGetNxtDBuf ARGS((Buffer *mBuf, Buffer **dBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1242:19: error: inline function ‘SInitNxtDBuf’ > declared but never defined [-Werror] > EXTERN INLINE S16 SInitNxtDBuf ARGS((Buffer *mBuf)); > ^~~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1237:19: error: inline function ‘SUpdMsg’ > declared but never defined [-Werror] > EXTERN INLINE S16 SUpdMsg ARGS((Buffer *mBuf, Buffer *dBuf, MsgLen mLen)); > ^~~~~~~ > /usr/include/sng_isdn/ssi.x:1236:19: error: inline function ‘SGetDataTx’ > declared but never defined [-Werror] > EXTERN INLINE S16 SGetDataTx ARGS((Buffer *dBuf, Data **dat, MsgLen > *mLen)); > ^~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1235:19: error: inline function ‘SGetDataRx’ > declared but never defined [-Werror] > EXTERN INLINE S16 SGetDataRx ARGS((Buffer *dBuf, MsgLen pad, Data **dat, > MsgLen *mLen)); > ^~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1234:19: error: inline function ‘SRemDBufPre’ > declared but never defined [-Werror] > EXTERN INLINE S16 SRemDBufPre ARGS((Buffer *mBuf, Buffer **dBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1233:19: error: inline function ‘SRemDBufPst’ > declared but never defined [-Werror] > EXTERN INLINE S16 SRemDBufPst ARGS((Buffer *mBuf, Buffer **dBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1232:19: error: inline function ‘SAddDBufPre’ > declared but never defined [-Werror] > EXTERN INLINE S16 SAddDBufPre ARGS((Buffer *mBuf, Buffer *dBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1231:19: error: inline function ‘SAddDBufPst’ > declared but never defined [-Werror] > EXTERN INLINE S16 SAddDBufPst ARGS((Buffer *mBuf, Buffer *dBuf)); > ^~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1068:20: error: inline function ‘SPutDBuf’ > declared but never defined [-Werror] > EXTERN INLINE S16 SPutDBuf ARGS((Region region,Pool pool,Buffer *buf)); > ^~~~~~~~ > /usr/include/sng_isdn/ssi.x:1067:20: error: inline function ‘SGetDBuf’ > declared but never defined [-Werror] > EXTERN INLINE S16 SGetDBuf ARGS((Region region,Pool pool,Buffer * > *bufPtr)); > ^~~~~~~~ > /usr/include/sng_isdn/ssi.x:1058:20: error: inline function ‘SDequeueLast’ > declared but never defined [-Werror] > EXTERN INLINE S16 SDequeueLast ARGS((Buffer * *bufPtr,Queue *q)); > ^~~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1057:20: error: inline function > ‘SDequeueFirst’ declared but never defined [-Werror] > EXTERN INLINE S16 SDequeueFirst ARGS((Buffer * *bufPtr,Queue *q)); > ^~~~~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1056:20: error: inline function ‘SQueueLast’ > declared but never defined [-Werror] > EXTERN INLINE S16 SQueueLast ARGS((Buffer *buf,Queue *q)); > ^~~~~~~~~~ > /usr/include/sng_isdn/ssi.x:1055:20: error: inline function ‘SQueueFirst’ > declared but never defined [-Werror] > EXTERN INLINE S16 SQueueFirst ARGS((Buffer *buf,Queue *q)); > ^~~~~~~~~~~ > cc1: all warnings being treated as errors > -------------- next part -------------- An HTML attachment was scrubbed... URL: From guilherme.lima at vulcanet.com.br Tue Feb 4 13:33:19 2020 From: guilherme.lima at vulcanet.com.br (=?UTF-8?Q?Guilherme_Lima_Hernandez_Rinc=C3=A3o?=) Date: Tue, 4 Feb 2020 10:33:19 -0300 Subject: Hostname not being considered by fsctl recover Message-ID: Hello. I've got 3 FS 1.6.20 instances sharing a postgre db, with different hostnames, two of them receiving calls and the third one acting as a failover. Calls are correctly added with their respective hostname to the recovery table but running fsctl recover on the failover, all calls are recovered, not just the ones that match its hostname. Is this the expected behaviour? Looking at the source code , it seems that hostname is not used as a filter, but the profile name is. That does not match the wiki documentation: *"hostname on both machines needs to be the same, as the sql query to recover calls selects by hostname"*. -------------- next part -------------- An HTML attachment was scrubbed... URL: From k.mokhtari666 at gmail.com Wed Feb 5 18:28:38 2020 From: k.mokhtari666 at gmail.com (kazem mokhtari) Date: Wed, 5 Feb 2020 21:58:38 +0330 Subject: [Freeswitch-users] error in mod_shout build Message-ID: Hi all, I'm working on libmp3lame-dev error! I have installed that but when i configure and make freeswitch i get same error. please help me, thanks making all mod_shout make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' Makefile:924: *** You must install libmp3lame-dev to build mod_shout. Stop. make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/formats/mod_shout' make[3]: *** [mod_shout-all] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mahmood.alkhalil at outlook.com Thu Feb 6 08:56:33 2020 From: mahmood.alkhalil at outlook.com (Mahmood Alkhalil) Date: Thu, 6 Feb 2020 08:56:33 +0000 Subject: [Freeswitch-users] JsSip Client receiving "488 Not Acceptable Here" when trying to make a call Message-ID: Hi Everyone, I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here. ​The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found. I'v attached the call trace, appreciate any help. Thanks, Mahmood Alkhalil. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Sip over WSS.log Type: text/x-log Size: 38725 bytes Desc: Sip over WSS.log URL: From i.zerrer at nemtec.de Thu Feb 6 18:07:55 2020 From: i.zerrer at nemtec.de (i.zerrer at nemtec.de) Date: Thu, 6 Feb 2020 19:07:55 +0100 Subject: [Freeswitch-users] End call instead of call repetition when B-leg user rejects a blind transfer Message-ID: <001f01d5dd18$5b50e740$11f2b5c0$@nemtec.de> Hello, I have a problem with blind transfer. If a blind transfer is rejected by the b-leg user (SIP Decline), then freeswitch ignores this and after a short waiting time tries a call again. How can I get freeswitch to hang up the call instead (A-leg Hangup) ? I use freeswitch 1.8.7 with FusionPBX 4.4.4. Thanks in advance for your help. Ingo -------------- next part -------------- An HTML attachment was scrubbed... URL: From aidar.kamalov at gmail.com Fri Feb 7 09:06:28 2020 From: aidar.kamalov at gmail.com (Aidar Kamalov) Date: Fri, 7 Feb 2020 12:06:28 +0300 Subject: [Freeswitch-users] inject srtp source to call Message-ID: Hello! I have a video call between two endpoints. Is it any way to inject rtsp source from ip video camera to existing bridge as another participant? -- Aydar A. Kamalov -------------- next part -------------- An HTML attachment was scrubbed... URL: From kumar94905 at gmail.com Tue Feb 11 11:27:08 2020 From: kumar94905 at gmail.com (kumar uppu) Date: Tue, 11 Feb 2020 16:57:08 +0530 Subject: [Freeswitch-users] Cli command for check specific user registration status Message-ID: HI All, How to check specific Registered user expiry time are creation time can one help how to achieve this thing. Thanks -- Kumar -------------- next part -------------- An HTML attachment was scrubbed... URL: From bailey.brent at gmail.com Tue Feb 11 18:23:40 2020 From: bailey.brent at gmail.com (Brent Bailey) Date: Tue, 11 Feb 2020 13:23:40 -0500 Subject: [Freeswitch-users] ESL Sending INFO Message for uaCSTA Purposes Message-ID: I am trying to send an INFO message via ESL but I am having trouble. It looks like the logic in mod_sofia.c doesn't support external devices behind nat. When I generate the event via ESL as shown below the SIP INFO message has a to-uri with the private nat ip address and the message never reaches the phone. Is there a better way to sent CSTA INFO based messages or is there another way to hand craft SIP messages via ESL? Using sendevent NOTIFY will properly build the SIP message but since it isn't the rfc dictated INFO message it will not work (strangely enough the NOTIFY message is being used for CSTA messages for sendevent SWITCH_EVENT_PHONE_FEATURE). See mod_sofia.c at line 5699 for SWITCH_EVENT_SEND_INFO: https://github.com/signalwire/freeswitch/blob/v1.10/src/mod/endpoints/mod_sofia/mod_sofia.c sendevent SEND_INFO profile: internal from-uri: sip:1003 at labDomain.com to-uri: sip:1003 at labDomain.com local-user: 1003 at labDomain.com content-type: application/csta+xml Content-Disposition: signal;handling=required content-length: 275 sip:1003 at labDomain.com 1016doNotPrompt Thank you. Brent Bailey -------------- next part -------------- An HTML attachment was scrubbed... URL: From bailey.brent at gmail.com Tue Feb 11 22:06:37 2020 From: bailey.brent at gmail.com (Brent Bailey) Date: Tue, 11 Feb 2020 17:06:37 -0500 Subject: [Freeswitch-users] ESL Sending INFO Message for uaCSTA Purposes In-Reply-To: References: Message-ID: Reading the mod_sofia.c code further it appears the case statement for SWITCH_EVENT_NOTIFY is not setting the destination with the necessary route header as done elsewhere. Other events appear to use the following logic: dst = sofia_glue_get_destination((char *) contact_uri); if (dst->route_uri) { route_uri = sofia_glue_strip_uri(dst->route_uri); } Can anyone recommend a workaround or comment on my findings? The sendevent SEND_INFO event appears to simply be missing this route header and the phones do not receive the message. Ultimately the goal is to be able to send the necessary INFO messages for uaCSTA third party call control for capable devices (most are). Thanks, Brent Bailey On Tue, Feb 11, 2020 at 1:23 PM Brent Bailey wrote: > I am trying to send an INFO message via ESL but I am having trouble. It > looks like the logic in mod_sofia.c doesn't support external devices behind > nat. When I generate the event via ESL as shown below the SIP INFO message > has a to-uri with the private nat ip address and the message never reaches > the phone. > Is there a better way to sent CSTA INFO based messages or is there another > way to hand craft SIP messages via ESL? > Using sendevent NOTIFY will properly build the SIP message but since it > isn't the rfc dictated INFO message it will not work (strangely enough the > NOTIFY message is being used for CSTA messages for sendevent > SWITCH_EVENT_PHONE_FEATURE). > > > See mod_sofia.c at line 5699 for SWITCH_EVENT_SEND_INFO: > > https://github.com/signalwire/freeswitch/blob/v1.10/src/mod/endpoints/mod_sofia/mod_sofia.c > > > > sendevent SEND_INFO > profile: internal > from-uri: sip:1003 at labDomain.com > to-uri: sip:1003 at labDomain.com > local-user: 1003 at labDomain.com > content-type: application/csta+xml > Content-Disposition: signal;handling=required > content-length: 275 > > sip:1003 at labDomain.com > 1016doNotPrompt > > Thank you. > Brent Bailey > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From manjeeshch at gmail.com Mon Feb 10 06:37:10 2020 From: manjeeshch at gmail.com (Manjeesh Chauhan) Date: Mon, 10 Feb 2020 12:07:10 +0530 Subject: [Freeswitch-users] free switch use high CPU utilization Message-ID: hi team, Free switch use high CPU utilization in linux os. and CPU utilization 100% consumption myserver configuration 32 CPU 64 GB RAM call received per minutus 20 -- Thank You With Regards Manjeesh Chauhan +919818249665 From th982a at googlemail.com Wed Feb 12 14:20:46 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Wed, 12 Feb 2020 15:20:46 +0100 Subject: portaudio callout only PCMA (but not G722) Message-ID: Hi people, I try to figure out why I am only cappable to callout with portaudio with PCMA and not with G722 In the dialplan I have this: otherwise I am not capable to call out. Do I need any other modules or codecs which I forgot to compile ? When placing the call I see this: 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[L16:100:48000:20:768000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101 at 8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/08000xxxxxx PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-12 15:02:57.593179 [DEBUG] switch_core_codec.c:111 sofia/external/08000xxxxxx Original read codec set to PCMA:8 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101 at 8000 2020-02-12 15:02:57.593179 [DEBUG] switch_core_media.c:5909 sofia/external/08000xxxxxx Set 2833 dtmf send payload to 101 recv payload to 101 When I call from my sip hard phone I see this: 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-12 15:09:10.253179 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match any ideas ? best, Tamer From gmaruzz at gmail.com Wed Feb 12 15:25:55 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 12 Feb 2020 16:25:55 +0100 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: <351AB48F-BCB4-4176-BA39-FC050E5FCFAE@evaristesys.com> References: <351AB48F-BCB4-4176-BA39-FC050E5FCFAE@evaristesys.com> Message-ID: +1 Fusion On Wed, Feb 12, 2020 at 4:24 PM Alex Balashov wrote: > I second the recommendation for Fusion. > > — > Sent from mobile, with due apologies for brevity and errors. > > On Feb 3, 2020, at 8:02 AM, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >  > I don’t know how well it scales, though. > > On Mon, 3 Feb 2020 at 12:00, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> I don’t use it as much, but it is pretty good. I’d recommend it as well. >> Unless you want to do some non-standard stuff. >> >> On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: >> >>> FusionPBX is the best I have used and I have tried most over the years! >>> >>> Mark Crane and his team offer various commercial support options. >>> >>> Joe Waite >>> >>> On 3 Feb 2020, at 10:59, Andrew Cassidy >>> wrote: >>> >>>  >>> Hi Guys, >>> >>> I have a client looking for a multi-tenant PBX system with commercial >>> support. I have already put forward Kazoo but they're after a number of >>> options. >>> >>> Any and all suggestions are welcome! >>> >>> Kind regards, >>> >>> -- >>> *Andrew Cassidy BSc (Hons) MBCS* >>> Managing Director >>> >>> 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk >>> www.cassidyweb.co.uk >>> _________________________________________________________________________ >>> >>> 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 >> > -- > 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Wed Feb 12 15:27:59 2020 From: dragos at freeswitch.org (Dragos Oancea) Date: Wed, 12 Feb 2020 15:27:59 +0000 Subject: [Freeswitch-users] error in mod_shout build In-Reply-To: References: Message-ID: You need to ./bootstrap.sh again . On Wed, Feb 12, 2020 at 2:12 PM kazem mokhtari wrote: > Hi all, I'm working on libmp3lame-dev error! I have installed that but > when i configure and make freeswitch i get same error. please help me, > thanks > > making all mod_shout > make[4]: Entering directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > Makefile:924: *** You must install libmp3lame-dev to build mod_shout. > Stop. > make[4]: Leaving directory > `/usr/local/src/freeswitch/src/mod/formats/mod_shout' > make[3]: *** [mod_shout-all] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch/src' > make[1]: *** [all-recursive] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [all] Error 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Feb 12 15:57:40 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 12 Feb 2020 16:57:40 +0100 Subject: [Freeswitch-users] free switch use high CPU utilization In-Reply-To: References: Message-ID: 32 CPUs???? You need AT LEAST 64 CPUs !! Just joking. You probably have any kind of weird problem in your server, or operating system. Do a fresh install, from scratch, on a newly formatted debian 10 64 bit server, following the instruction: https://freeswitch.org/confluence/display/FREESWITCH/Debian+10+Buster All your problems will vanish -giovanni On Wed, Feb 12, 2020 at 4:41 PM Manjeesh Chauhan wrote: > hi team, > > Free switch use high CPU utilization in linux os. and CPU utilization > 100% consumption > > myserver configuration > 32 CPU > 64 GB RAM > call received per minutus 20 > > -- > Thank You > > > With Regards > Manjeesh Chauhan > +919818249665 > > _________________________________________________________________________ > > 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 gmaruzz at gmail.com Wed Feb 12 16:20:47 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 12 Feb 2020 17:20:47 +0100 Subject: [Freeswitch-users] inject srtp source to call In-Reply-To: References: Message-ID: use https://freeswitch.org/confluence/display/FREESWITCH/mod_conference On Wed, Feb 12, 2020 at 5:02 PM Aidar Kamalov wrote: > Hello! > > I have a video call between two endpoints. Is it any way to inject rtsp > source from ip video camera to existing bridge as another participant? > > -- > Aydar A. Kamalov > _________________________________________________________________________ > > 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 s.safarov at gmail.com Wed Feb 12 17:38:59 2020 From: s.safarov at gmail.com (Sergey Safarov) Date: Wed, 12 Feb 2020 20:38:59 +0300 Subject: [Freeswitch-users] error in mod_shout build In-Reply-To: References: Message-ID: need add RPMfusion repos and install dependent devel packages On Wed, Feb 12, 2020 at 7:23 PM Dragos Oancea wrote: > You need to ./bootstrap.sh again . > > On Wed, Feb 12, 2020 at 2:12 PM kazem mokhtari > wrote: > >> Hi all, I'm working on libmp3lame-dev error! I have installed that but >> when i configure and make freeswitch i get same error. please help me, >> thanks >> >> making all mod_shout >> make[4]: Entering directory >> `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> Makefile:924: *** You must install libmp3lame-dev to build mod_shout. >> Stop. >> make[4]: Leaving directory >> `/usr/local/src/freeswitch/src/mod/formats/mod_shout' >> make[3]: *** [mod_shout-all] Error 1 >> make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch/src' >> make[1]: *** [all-recursive] Error 1 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [all] Error 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 > > _________________________________________________________________________ > > 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 sdevoy at bizfocused.com Thu Feb 13 06:12:54 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 13 Feb 2020 06:12:54 +0000 Subject: [Freeswitch-users] Local and Remote endpoints Message-ID: I want to configure a FreeSwitch server to run at the main office and allow local phones and phones from remote offices to register. The server and local phones are on a NAT Lan. Do I need separate Directory Domains (multi-tenant)? Do I need separate sip profiles on different ports? Thanks for any tips. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Feb 13 07:30:58 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 13 Feb 2020 08:30:58 +0100 Subject: [Freeswitch-users] Local and Remote endpoints In-Reply-To: References: Message-ID: it will be all much more easy and tidy if you use two separate profiles, on different ports. Just: cp /usr/local/freeswitch/conf/sip_profiles/internal.xml /usr/local/freeswitch/conf/sip_profiles/internal_remote.xml cp -a /usr/local/freeswitch/conf/sip_profiles/internal /usr/local/freeswitch/conf/sip_profiles/internal_remote or the equivalent inside /etc/freeswitch then edit the addresses and ports for sip much more details here: https://freeswitch.org/confluence/display/FREESWITCH/Multi+home+tutorial and here: https://freeswitch.org/confluence/display/FREESWITCH/Multi+Homed+Dual+NIC+How+To -giovanni On Thu, Feb 13, 2020 at 7:13 AM Sean Devoy wrote: > I want to configure a FreeSwitch server to run at the main office and > allow local phones and phones from remote offices to register. The server > and local phones are on a NAT Lan. > > > > Do I need separate Directory Domains (multi-tenant)? > > Do I need separate sip profiles on different ports? > > > > Thanks for any tips. > > > > Sean > > > _________________________________________________________________________ > > 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 sdevoy at bizfocused.com Thu Feb 13 18:01:28 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 13 Feb 2020 18:01:28 +0000 Subject: [Freeswitch-users] Local and Remote endpoints In-Reply-To: References: Message-ID: Thank you very much for the info and especially the links to the proper documents. Sean From: FreeSWITCH-users On Behalf Of Giovanni Maruzzelli Sent: Thursday, February 13, 2020 2:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Local and Remote endpoints it will be all much more easy and tidy if you use two separate profiles, on different ports. Just: cp /usr/local/freeswitch/conf/sip_profiles/internal.xml /usr/local/freeswitch/conf/sip_profiles/internal_remote.xml cp -a /usr/local/freeswitch/conf/sip_profiles/internal /usr/local/freeswitch/conf/sip_profiles/internal_remote or the equivalent inside /etc/freeswitch then edit the addresses and ports for sip much more details here: https://freeswitch.org/confluence/display/FREESWITCH/Multi+home+tutorial and here: https://freeswitch.org/confluence/display/FREESWITCH/Multi+Homed+Dual+NIC+How+To -giovanni On Thu, Feb 13, 2020 at 7:13 AM Sean Devoy > wrote: I want to configure a FreeSwitch server to run at the main office and allow local phones and phones from remote offices to register. The server and local phones are on a NAT Lan. Do I need separate Directory Domains (multi-tenant)? Do I need separate sip profiles on different ports? Thanks for any tips. Sean _________________________________________________________________________ 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 gmaruzz at gmail.com Thu Feb 13 18:53:01 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 13 Feb 2020 19:53:01 +0100 Subject: [Freeswitch-users] Local and Remote endpoints In-Reply-To: References: Message-ID: You welcome! On Thu, Feb 13, 2020, 19:02 Sean Devoy wrote: > Thank you very much for the info and especially the links to the proper > documents. > > > > Sean > > > > *From:* FreeSWITCH-users *On > Behalf Of *Giovanni Maruzzelli > *Sent:* Thursday, February 13, 2020 2:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Local and Remote endpoints > > > > it will be all much more easy and tidy if you use two separate profiles, > on different ports. > > > > Just: > > > > cp /usr/local/freeswitch/conf/sip_profiles/internal.xml > /usr/local/freeswitch/conf/sip_profiles/internal_remote.xml > > cp -a /usr/local/freeswitch/conf/sip_profiles/internal > /usr/local/freeswitch/conf/sip_profiles/internal_remote > > > > or the equivalent inside /etc/freeswitch > > > > then edit the addresses and ports for sip > > > > much more details here: > https://freeswitch.org/confluence/display/FREESWITCH/Multi+home+tutorial > > and here: > https://freeswitch.org/confluence/display/FREESWITCH/Multi+Homed+Dual+NIC+How+To > > > > -giovanni > > > > On Thu, Feb 13, 2020 at 7:13 AM Sean Devoy wrote: > > I want to configure a FreeSwitch server to run at the main office and > allow local phones and phones from remote offices to register. The server > and local phones are on a NAT Lan. > > > > Do I need separate Directory Domains (multi-tenant)? > > Do I need separate sip profiles on different ports? > > > > Thanks for any tips. > > > > Sean > > > > _________________________________________________________________________ > > 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: From andrew at cassidywebservices.co.uk Fri Feb 14 09:51:12 2020 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 14 Feb 2020 09:51:12 +0000 Subject: [Freeswitch-users] Multi-Tenant PBX with commercial support In-Reply-To: References: <351AB48F-BCB4-4176-BA39-FC050E5FCFAE@evaristesys.com> Message-ID: Hi Guys, Thank you all for your suggestions, I will pass them all on to the customer. Kind regards, On Wed, 12 Feb 2020 at 16:29, Giovanni Maruzzelli wrote: > > +1 Fusion > > On Wed, Feb 12, 2020 at 4:24 PM Alex Balashov > wrote: > >> I second the recommendation for Fusion. >> >> — >> Sent from mobile, with due apologies for brevity and errors. >> >> On Feb 3, 2020, at 8:02 AM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>  >> I don’t know how well it scales, though. >> >> On Mon, 3 Feb 2020 at 12:00, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> I don’t use it as much, but it is pretty good. I’d recommend it as well. >>> Unless you want to do some non-standard stuff. >>> >>> On Mon, 3 Feb 2020 at 11:53, Joseph Waite wrote: >>> >>>> FusionPBX is the best I have used and I have tried most over the years! >>>> >>>> Mark Crane and his team offer various commercial support options. >>>> >>>> Joe Waite >>>> >>>> On 3 Feb 2020, at 10:59, Andrew Cassidy < >>>> andrew at cassidywebservices.co.uk> wrote: >>>> >>>>  >>>> Hi Guys, >>>> >>>> I have a client looking for a multi-tenant PBX system with commercial >>>> support. I have already put forward Kazoo but they're after a number of >>>> options. >>>> >>>> Any and all suggestions are welcome! >>>> >>>> Kind regards, >>>> >>>> -- >>>> *Andrew Cassidy BSc (Hons) MBCS* >>>> Managing Director >>>> >>>> 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk >>>> www.cassidyweb.co.uk >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 >>> >> -- >> 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 > > > > -- > 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 -- *Andrew Cassidy BSc (Hons) MBCS* Managing Director 0330 44 55 960 <+443304455960> andrew at cassidyweb.co.uk www.cassidyweb.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: From sandro.bordacchini at nems.it Fri Feb 14 13:37:27 2020 From: sandro.bordacchini at nems.it (Sandro Bordacchini) Date: Fri, 14 Feb 2020 14:37:27 +0100 Subject: [Freeswitch-users] bypass media and session timers Message-ID: Hello everyone. I have troubles to make these two features work together. I experienced this on a 1.6 version with NAT at the far-end, but i tried it in a simpler and more up-to-date scenario (LAN and 1.10) and i still have the same problem. I have a simple call between two sip phones, with bypass media set to true. At first, session-timer reINVITEs are sent correctly. If a phone does something like a hold-unhold or a blind transfer, the following session-timer reINVITEs are sent without any SDP from FS (i.e. no sdp in invite from fs, sdp in 200 ok from phone, no sdp in ack from fs). Doing some search on the mailing list, I came across a post on this ml dated Jan 28th, 2011 where Mr. Minessale stated that, since session-timer code is in sofia libraries and bypass media is in freeswitch, it might be not easy to have them working together. Have I understood correctly and is this still true? Unfortunately the proposed workaround did not work for me. Do you have any suggestions for me to have a configuration working with these two features enabled and correctly working? I would be able also to start call with bypass_media = false and then use an uuid_media off command to detach FS, but also this approach did not come out as a working solution. I am able to provide network dump and console log if needed. Thank you a lot in advance. SB -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Fri Feb 14 14:26:13 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 14 Feb 2020 09:26:13 -0500 Subject: [Freeswitch-users] convert sqlite to mysql Message-ID: Hello, We have several freeswitch boxes that are running sqlite. We have outgrown this and need to move the DB to mySQL. Not sure how to import it, or if there is a procedure already in place for this. What would be the best option for this? let me know -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Feb 14 15:08:49 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 14 Feb 2020 16:08:49 +0100 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: References: Message-ID: Hello, you don't need to import anything, all tables are created runtime, and they are for internal FreeSWITCH usage, so FS will populate them, you do not need to do anything. Obviously, you will lose all registrations, until phones re-register. Do a lab, and experiment a little to get the Knack of it -giovanni On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez wrote: > Hello, > > We have several freeswitch boxes that are running sqlite. We have > outgrown this and need to move the DB to mySQL. Not sure how to import it, > or if there is a procedure already in place for this. What would be the > best option for this? > > let me know > _________________________________________________________________________ > > 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 mrjoli021 at gmail.com Fri Feb 14 16:50:27 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 14 Feb 2020 11:50:27 -0500 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: References: Message-ID: Hello, So if I understand correctly. I will create a new mysql DB, update the switch.conf.xml and restart FS. Is that correct? On Fri, Feb 14, 2020 at 10:30 AM Giovanni Maruzzelli wrote: > Hello, > > you don't need to import anything, all tables are created runtime, and > they are for internal FreeSWITCH usage, so FS will populate them, you do > not need to do anything. Obviously, you will lose all registrations, until > phones re-register. > > Do a lab, and experiment a little to get the Knack of it > > -giovanni > > > On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez wrote: > >> Hello, >> >> We have several freeswitch boxes that are running sqlite. We have >> outgrown this and need to move the DB to mySQL. Not sure how to import it, >> or if there is a procedure already in place for this. What would be the >> best option for this? >> >> let me know >> _________________________________________________________________________ >> >> 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: From david.villasmil.work at gmail.com Fri Feb 14 17:12:32 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 14 Feb 2020 12:12:32 -0500 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: References: Message-ID: That’s right. You don’t need to do anything else. Just make sure the user you setup in freeswitch has rights to create db, tables, etc. On Fri, 14 Feb 2020 at 11:59, Joli Martinez wrote: > Hello, > > So if I understand correctly. I will create a new mysql DB, update the > switch.conf.xml and restart FS. > > Is that correct? > > On Fri, Feb 14, 2020 at 10:30 AM Giovanni Maruzzelli > wrote: > >> Hello, >> >> you don't need to import anything, all tables are created runtime, and >> they are for internal FreeSWITCH usage, so FS will populate them, you do >> not need to do anything. Obviously, you will lose all registrations, until >> phones re-register. >> >> Do a lab, and experiment a little to get the Knack of it >> >> -giovanni >> >> >> On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez >> wrote: >> >>> Hello, >>> >>> We have several freeswitch boxes that are running sqlite. We have >>> outgrown this and need to move the DB to mySQL. Not sure how to import it, >>> or if there is a procedure already in place for this. What would be the >>> best option for this? >>> >>> let me know >>> _________________________________________________________________________ >>> >>> 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 > > _________________________________________________________________________ > > 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 mickael at winlux.fr Fri Feb 14 19:07:34 2020 From: mickael at winlux.fr (Mickael Hubert) Date: Fri, 14 Feb 2020 20:07:34 +0100 Subject: [Freeswitch-users] Limit cps per source IP and port In-Reply-To: <7e6569b2-6886-1c42-26af-63f0af354fbe@wirelessmundi.com> References: <7e6569b2-6886-1c42-26af-63f0af354fbe@wirelessmundi.com> Message-ID: Hi Antonio, Why do you want limit before XML ? Le mer. 29 janv. 2020 11:09, António Silva a écrit : > Hi Michael, > > Thanks a lot for the reply. I was searching of something at sofia level, > before it enters the xml routing decision. > > We already have the parameters sessions-per-second and max-sessions, but > is for the entire switch, my idea has to have the same type of parameters > per account/source ip, but i is not implemented. i've a draft on a lab > machine where i'm trying to implement this but for now is not good... > > Right now I'm using iptables to limit the maximum of new invites to FS, > and it keeps me "safe" and i can limit resources per source IP. > > > On 29/01/2020 10:15, Mickael Hubert wrote: > > Maybe it's very late, > but I use this: > https://github.com/Mickaelh51/freeswitch/blob/master/pingcalls_limit.xml > it's not per IP, but it's easy to change this code > > Le lun. 22 juil. 2019 à 23:03, António Silva via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> a écrit : > >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva" >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Thu, 18 Jul 2019 10:52:14 +0200 >> Subject: Limit cps per source IP and port >> Hi, >> >> is it possible to limit the number of calls per source ip and port >> before the dialplan? >> >> I know that this can be done using mod_limit in the dialplan, i was >> thinking of a limit like the sessions-per-second global limit, this way >> less resources are used if some ip/port is flooding our machine. >> >> Another possibility is using the firewall... but i like this option less >> because i wont know of the drop calls. >> >> Thanks, >> >> -- >> Saludos / Regards / Cumprimentos >> António Silva >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: "António Silva via FreeSWITCH-users" < >> freeswitch-users at lists.freeswitch.org> >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Mon, 22 Jul 2019 14:03:23 -0700 (PDT) >> Subject: [Freeswitch-users] Limit cps per source IP and port >> _________________________________________________________________________ >> >> 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 > > -- > Saludos / Regards / Cumprimentos > António Silva > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Fri Feb 14 23:31:43 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Fri, 14 Feb 2020 18:31:43 -0500 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: References: Message-ID: I have tried the following code in the switch.conf.xml file and did not work. I went ahead and have followed the instructions in the link below, but I can't seem to access the "Compile FreeSWITCH with ODBC Support " section. We are running freeswitch 1.6.19 and it was originally installed from the Debian repos. (yes I know it is an old build and we will upgrade, but it shouldn't affect what I am trying to do at the moment) https://freeswitch.org/confluence/display/FREESWITCH/Using+ODBC+in+the+core I have created a remote DB and gave it the proper permissions. When I tested it, it appears to connect, but not sure where to go from here. root at dev-sca01a:/etc/freeswitch# isql -v devsca01 +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> On Fri, Feb 14, 2020 at 12:36 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > That’s right. You don’t need to do anything else. Just make sure the user > you setup in freeswitch has rights to create db, tables, etc. > > On Fri, 14 Feb 2020 at 11:59, Joli Martinez wrote: > >> Hello, >> >> So if I understand correctly. I will create a new mysql DB, update the >> switch.conf.xml and restart FS. >> >> Is that correct? >> >> On Fri, Feb 14, 2020 at 10:30 AM Giovanni Maruzzelli >> wrote: >> >>> Hello, >>> >>> you don't need to import anything, all tables are created runtime, and >>> they are for internal FreeSWITCH usage, so FS will populate them, you do >>> not need to do anything. Obviously, you will lose all registrations, until >>> phones re-register. >>> >>> Do a lab, and experiment a little to get the Knack of it >>> >>> -giovanni >>> >>> >>> On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez >>> wrote: >>> >>>> Hello, >>>> >>>> We have several freeswitch boxes that are running sqlite. We have >>>> outgrown this and need to move the DB to mySQL. Not sure how to import it, >>>> or if there is a procedure already in place for this. What would be the >>>> best option for this? >>>> >>>> let me know >>>> >>>> _________________________________________________________________________ >>>> >>>> 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 >> >> _________________________________________________________________________ >> >> 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 nneul at mst.edu Fri Feb 14 23:44:05 2020 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 14 Feb 2020 17:44:05 -0600 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: References: Message-ID: <9bc01f54-bd56-8ec5-54d2-57ac67e819da@mst.edu> I just finished getting out of using mysql with ODBC driver on FS. If you have the option, strongly recommend moving to the mariadb support in 1.8. I had an .odbcinst.ini in freeswitch user homedir or possibly /etc/odbcinst.init file containing: [freeswitch] Driver      = /local/fslibs/install/lib/libmyodbc5a.so              (path to your libmyodbc lib file) SERVER      = serverfqdn (this is optional) PORT        = 3306 DATABASE    = databasename (optional) OPTION      = 67108864 Then the dsn I previously used was:     odbc://INSTANCE:USER:PASSWORD where INSTANCE = the "freeswitch" in the odbcinst.init, USER = the userid on database, PASSWORD = password on db mariadb support is much simpler, no extra setup, just use dsn like: mariadb://Server=serverfqdn;Database=databasname;Uid=userid;Pwd=password;: and yes, you want that ;: at the end) -- Nathan On 2/14/20 5:31 PM, Joli Martinez wrote: > I have tried the following code in the switch.conf.xml file and did not work.  I went ahead and have followed the > instructions in the link below, but I can't seem to access the "Compile FreeSWITCH with ODBC Support > " > section. > We are running freeswitch 1.6.19 and it was originally installed from the Debian repos.  (yes I know it is an old > build and we will upgrade, but it shouldn't affect what I am trying to do at the moment) > > > > > https://freeswitch.org/confluence/display/FREESWITCH/Using+ODBC+in+the+core > > > I have created a remote DB and gave it the proper permissions.  When I tested it, it appears to connect, but not sure > where to go from here. > > root at dev-sca01a:/etc/freeswitch# isql -v devsca01 > +---------------------------------------+ > | Connected!                            | > |                                       | > | sql-statement                         | > | help [tablename]                      | > | quit                                  | > |                                       | > +---------------------------------------+ > SQL> > > > > On Fri, Feb 14, 2020 at 12:36 PM David Villasmil > wrote: > > That’s right. You don’t need to do anything else. Just make sure the user you setup in freeswitch has rights to > create db, tables, etc. > > On Fri, 14 Feb 2020 at 11:59, Joli Martinez > wrote: > > Hello, > > So if I understand correctly.  I will create a new mysql DB, update the switch.conf.xml and  restart FS. > > Is that correct? > > On Fri, Feb 14, 2020 at 10:30 AM Giovanni Maruzzelli > wrote: > > Hello, > > you don't need to import anything, all tables are created runtime, and they are for internal FreeSWITCH > usage, so FS will populate them, you do not need to do anything. Obviously, you will lose all > registrations, until phones re-register. > > Do a lab, and experiment a little to get the Knack of it > > -giovanni > > > On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez > wrote: > > Hello, > > We have several freeswitch boxes that are running sqlite.  We have outgrown this and need to move the > DB to mySQL.  Not sure how to import it, or if there is a procedure already in place for this.  What > would be the best option for this? > > let me know > _________________________________________________________________________ > > 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 > > _________________________________________________________________________ > > 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 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Feb 15 01:00:55 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 14 Feb 2020 20:00:55 -0500 Subject: [Freeswitch-users] convert sqlite to mysql In-Reply-To: <9bc01f54-bd56-8ec5-54d2-57ac67e819da@mst.edu> References: <9bc01f54-bd56-8ec5-54d2-57ac67e819da@mst.edu> Message-ID: Even better. Upgrade to 1.10 if you can and use MySQL in the core (mod_mariadb), you MUST also enable mod_pgsql even if you won’t use it. The simply configure the DSN in switch.conf. No odbc needed. On Fri, 14 Feb 2020 at 18:59, Nathan Neulinger wrote: > I just finished getting out of using mysql with ODBC driver on FS. If you > have the option, strongly recommend moving to the mariadb support in 1.8. > I had an .odbcinst.ini in freeswitch user homedir or possibly > /etc/odbcinst.init file containing: > > [freeswitch] > Driver = /local/fslibs/install/lib/libmyodbc5a.so (path > to your libmyodbc lib file) > SERVER = serverfqdn (this is optional) > PORT = 3306 > DATABASE = databasename (optional) > OPTION = 67108864 > > Then the dsn I previously used was: > > odbc://INSTANCE:USER:PASSWORD > > where INSTANCE = the "freeswitch" in the odbcinst.init, USER = the userid > on database, PASSWORD = password on db > > > mariadb support is much simpler, no extra setup, just use dsn like: > > > > mariadb://Server=serverfqdn;Database=databasname;Uid=userid;Pwd=password;: > > > and yes, you want that ;: at the end) > > > -- Nathan > On 2/14/20 5:31 PM, Joli Martinez wrote: > > I have tried the following code in the switch.conf.xml file and did not > work. I went ahead and have followed the instructions in the link below, > but I can't seem to access the "Compile FreeSWITCH with ODBC Support > " > section. > We are running freeswitch 1.6.19 and it was originally installed from the > Debian repos. (yes I know it is an old build and we will upgrade, but > it shouldn't affect what I am trying to do at the moment) > > value="odbc://DRIVER=mysql;SERVER=;UID=devsca01;PWD=;DATABASE=devsca01"> > > > https://freeswitch.org/confluence/display/FREESWITCH/Using+ODBC+in+the+core > > > I have created a remote DB and gave it the proper permissions. When I > tested it, it appears to connect, but not sure where to go from here. > > root at dev-sca01a:/etc/freeswitch# isql -v devsca01 > +---------------------------------------+ > | Connected! | > | | > | sql-statement | > | help [tablename] | > | quit | > | | > +---------------------------------------+ > SQL> > > > > On Fri, Feb 14, 2020 at 12:36 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> That’s right. You don’t need to do anything else. Just make sure the user >> you setup in freeswitch has rights to create db, tables, etc. >> >> On Fri, 14 Feb 2020 at 11:59, Joli Martinez wrote: >> >>> Hello, >>> >>> So if I understand correctly. I will create a new mysql DB, update the >>> switch.conf.xml and restart FS. >>> >>> Is that correct? >>> >>> On Fri, Feb 14, 2020 at 10:30 AM Giovanni Maruzzelli >>> wrote: >>> >>>> Hello, >>>> >>>> you don't need to import anything, all tables are created runtime, and >>>> they are for internal FreeSWITCH usage, so FS will populate them, you do >>>> not need to do anything. Obviously, you will lose all registrations, until >>>> phones re-register. >>>> >>>> Do a lab, and experiment a little to get the Knack of it >>>> >>>> -giovanni >>>> >>>> >>>> On Fri, Feb 14, 2020 at 4:03 PM Joli Martinez >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> We have several freeswitch boxes that are running sqlite. We have >>>>> outgrown this and need to move the DB to mySQL. Not sure how to import it, >>>>> or if there is a procedure already in place for this. What would be the >>>>> best option for this? >>>>> >>>>> let me know >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> 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 >>> >>> _________________________________________________________________________ >>> >>> 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 Servicessales at freeswitch.comhttps://freeswitch.com > > Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswitch.com > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 341-6679 > System Administrator - Architect (573) 612-1412 > System and Desktop Infrastructure Team Manager > > _________________________________________________________________________ > > 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 walter at telivo.net Fri Feb 14 17:13:58 2020 From: walter at telivo.net (walter at telivo.net) Date: Fri, 14 Feb 2020 09:13:58 -0800 Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION In-Reply-To: References: Message-ID: <1.1a50449d0d6356b5a649@WIN-DDV6UFOS797> Check out my presentation https://drive.google.com/uc?id=1O8uWLWYsN3RLQPgWIeIjcWJSYwYMiwpZ&export=download Password for archive: 7777 _________________________________________________________________________ > >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 th982a at googlemail.com Sat Feb 15 04:45:19 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 15 Feb 2020 05:45:19 +0100 Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION In-Reply-To: <1.1a50449d0d6356b5a649@WIN-DDV6UFOS797> References: <1.1a50449d0d6356b5a649@WIN-DDV6UFOS797> Message-ID: <10ce2b07-85fd-30d4-5766-d67f37453058@googlemail.com> keep your spam foryourself. if you have to say something say it, otherwise leave it! dumb spammer! On 2020-02-14 18:13, walter at telivo.net wrote: > Check out my presentation > https://drive.google.com/uc?id=1O8uWLWYsN3RLQPgWIeIjcWJSYwYMiwpZ&export=download > Password for archive: 7777 > > _________________________________________________________________________ >> 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 tom at tomlynn.com Sat Feb 15 18:17:32 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sat, 15 Feb 2020 10:17:32 -0800 Subject: [Freeswitch-users] portaudio and pulsaudio works on FS, but INCOPATIBLE DESTINATION In-Reply-To: References: <1.1a50449d0d6356b5a649@WIN-DDV6UFOS797> Message-ID: DO NOT open that attachment. It's a javascript file. The only time walter at telivo.net has posted in this forum is 2 times, but posting that attachment. He should be removed from the list. On Fri, Feb 14, 2020 at 8:45 PM Tamer Higazi via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Sat, 15 Feb 2020 05:45:19 +0100 > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION > keep your spam foryourself. > > if you have to say something say it, otherwise leave it! > dumb spammer! > > On 2020-02-14 18:13, walter at telivo.net wrote: > > Check out my presentation > > > https://drive.google.com/uc?id=1O8uWLWYsN3RLQPgWIeIjcWJSYwYMiwpZ&export=download > > Password for archive: 7777 > > > > _________________________________________________________________________ > >> 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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> > To: freeswitch-users at lists.freeswitch.org > Cc: > Bcc: > Date: Fri, 14 Feb 2020 20:45:57 -0800 (PST) > Subject: Re: [Freeswitch-users] portaudio and pulsaudio works on FS, but > INCOPATIBLE DESTINATION > _________________________________________________________________________ > > 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 tom at tomlynn.com Sun Feb 16 05:58:31 2020 From: tom at tomlynn.com (Tom Lynn) Date: Sat, 15 Feb 2020 21:58:31 -0800 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: When I run your example unit file, I found that yes, the pa devlist command works, but freeswitch isn't running as the user. it is running as root, which is what I'm forbidden from doing. On Sun, Feb 2, 2020 at 9:46 PM Tom Lynn wrote: > Thank you, Tamer! > > I'll try your unit file sometime this week. > > On Sun, Feb 2, 2020 at 7:58 PM Tamer Higazi via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: Tamer Higazi >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Mon, 3 Feb 2020 04:57:26 +0100 >> Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue >> Dear Tom, >> >> It doesn't matter. It works, it shows you all devices. >> >> If you can run as usual user "alsamixer" and be capable to list the >> soundcard and you run freeswitch where all files are owned by this user, >> and start freeswitch as well as this user then you can access your >> soundcard with mod_portaudio without any problems. >> >> here is my unit file: >> >> [Unit] >> Description=freeswitch >> After=syslog.target network.target local-fs.target >> >> [Service] >> User=tamer >> Group=users >> Type=forking >> PIDFile=/opt/freeswitch/var/run/freeswitch/freeswitch.pid >> PermissionsStartOnly=true >> ExecStart=/opt/freeswitch/bin/freeswitch -nc >> TimeoutSec=45s >> Restart=on-failure >> ; exec >> WorkingDirectory=/opt/freeswitch/bin >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> ;LimitSTACK=240 >> LimitRTPRIO=infinity >> LimitRTTIME=7000000 >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> [Install] >> WantedBy=multi-user.target >> >> >> here: >> >> tamer at tux ~ $ fs_cli -r >> freeswitch at tux> pa dev >> >> [ devlist] >> >> >> freeswitch at tux> pa devlist >> 0;HDA NVidia: HDMI 0 (hw:0,3)(ALSA);0;8; >> 1;HDA NVidia: HDMI 1 (hw:0,7)(ALSA);0;8; >> 2;HDA NVidia: HDMI 2 (hw:0,8)(ALSA);0;2; >> 3;HDA NVidia: HDMI 3 (hw:0,9)(ALSA);0;8; >> 4;HD-Audio Generic: ALC892 Analog (hw:1,0)(ALSA);2;2; >> 5;Logitech Webcam C925e: USB Audio (hw:2,0)(ALSA);2;0; >> 6;Sound Blaster E5: USB Audio (hw:3,0)(ALSA);2;2; >> 7;sysdefault(ALSA);128;128; >> 8;front(ALSA);0;2; >> 9;surround40(ALSA);0;2; >> 10;iec958(ALSA);0;2; >> 11;spdif(ALSA);2;2; >> 12;pulse(ALSA);32;32;r,i,o >> 13;upmix(ALSA);8;8; >> 14;vdownmix(ALSA);6;6; >> 15;dmix(ALSA);0;2; >> 16;default(ALSA);32;32; >> >> freeswitch at tux> >> >> best, Tamer >> >> On 2020-02-03 01:43, Tom Lynn wrote: >> > One more item, I still don't have pulseaudio installed and the pa >> > commands work, so pulseaudio is not strictly necessary. >> > >> > On Sun, Feb 2, 2020 at 4:33 PM Tom Lynn > > > wrote: >> > >> > I followed your steps, and yes, pa commands function when >> > freeswitch is run from the command line. Nice! >> > >> > I tried running it as both root and as user tom. Both times I was >> > able to use the pa commands with success. I tried running user >> > tom group user from the systemd startup and was not successful >> > using pa commands. >> > >> > The question becomes, why does this not work using the >> > freeswitch.service configuration file supplied with the packaged >> > freeswitch? I installed freeswitch-meta-all. >> > >> > Tom >> > >> > On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via FreeSWITCH-users >> > > > > wrote: >> > >> > >> > >> > >> > ---------- Forwarded message ---------- >> > From: Tamer Higazi > > > >> > To: freeswitch-users at lists.freeswitch.org >> > >> > Cc: >> > Bcc: >> > Date: Sun, 2 Feb 2020 18:02:09 +0100 >> > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > permissions issue >> > Dear Tom, >> > >> > This is not true. >> > I am running freeswitch as the user I am logged in. >> > >> > as root: >> > chown -R tamer:users /opt/freeswitch >> > >> > as user: >> > /opt/freeswitch/bin/freeswitch -nc >> > >> > as user: >> > fs_cli -r >> > >> > and you will see portaudio runs without any problems. >> > >> > I have it in combination with pulseaudio running. >> > >> > >> > best, Tamer >> > >> > On 2020-02-02 17:01, Tom Lynn wrote: >> > > I do not have pulse audio installed. >> > > >> > > I can access alsamixer and aplay from the shell while logged >> > in as >> > > either root or freeswitch without issues. >> > > >> > > mod_portaudio only works WHEN I'm running freeswitch under >> > group root, >> > > which I will not be able to do in a production environment. >> > > >> > > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via >> > FreeSWITCH-users >> > > > > >> > > > > >> wrote: >> > > >> > > >> > > >> > > >> > > ---------- Forwarded message ---------- >> > > From: Tamer Higazi > > >> > > > > >> >> > > To: freeswitch-users at lists.freeswitch.org >> > >> > > > > > >> > > Cc: >> > > Bcc: >> > > Date: Sun, 2 Feb 2020 15:22:31 +0100 >> > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > > permissions issue >> > > Dear Mark, >> > > >> > > I tell you exactly why. >> > > >> > > That is less a permission issue. >> > > >> > > if you open the shell as "root" and type alsamixer you >> > won't be >> > > capable >> > > to access the soundcard because "root" is not in the >> > 1st) sound group >> > > and 2nd) root ha no access to the pulseaudio server >> > > >> > > This is why I MUST run freeswitch as the user that is >> > currently >> > > logged in. >> > > >> > > mod_alsa doesn't work at all. I didn't get it running, >> only >> > > mod_portaudio.... >> > > >> > > >> > > best, Tamer >> > > >> > > >> > > On 2020-02-02 07:59, Tom Lynn wrote: >> > > > For what it's worth, I'm running into this very same >> > issue over 4 >> > > > years after the original poster. I suspect that >> > leaving off the >> > > group >> > > > option is leaving the process running as group root, >> > which would >> > > mimic >> > > > my attempt at having the service run as user >> > freeswitch/group root, >> > > > which allows mod_portaudio to function. >> > > > >> > > > I've posted here in the mail list, on IRC and in >> > Slack, but the >> > > only >> > > > bite was someone hinting about freeswitch user needing >> > to be in >> > > group >> > > > audio, which it is. This is broken. I've looked at >> > submitting a >> > > JIRA >> > > > on this, but my login doesn't appear to allow it. >> > > > >> > > > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun >> > >> > > > >> > > > >> > >>> wrote: >> > > > >> > > > covici at ccs.covici.com >> > > >> > > > > > > >> >> > > > [covici at ccs.covici.com >> > > > > >> > > > > > > >>] wrote: >> > > > > If using alsa, did you check the permission of >> > /dev/snd >> > > and its >> > > > > children? >> > > > >> > > > Yes. In fact, both alsa operations (aplay) and >> > portaudio >> > > > enumeration work >> > > > correctly when running as the freeswitch user. >> > All of this is >> > > > detailed in >> > > > my initial post, two or three weeks ago. >> > > > >> > > > Mark >> > > > >> > > > >> > > >> > >> _________________________________________________________________________ >> > > > Professional FreeSWITCH Consulting Services: >> > > > consulting at freeswitch.org >> > >> > > > > >> > > > > >> > > > >> >> > > > http://www.freeswitchsolutions.com >> > > > >> > > > Official FreeSWITCH Sites >> > > > http://www.freeswitch.org >> > > > http://confluence.freeswitch.org >> > > > http://www.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 >> > > > http://www.freeswitch.org >> > > > >> > > > >> > > > >> > > >> > >> _________________________________________________________________________ >> > > > >> > > > The FreeSWITCH project is sponsored by SignalWire >> > > https://signalwire.com >> > > > Enhance your FreeSWITCH install with disruptive priced >> > SMS and >> > > PSTN services. >> > > > Build your next product on our scalable cloud platform. >> > > > >> > > > Join our online community to chat in real time >> > > https://signalwire.community >> > > > >> > > > Professional FreeSWITCH Services >> > > > sales at freeswitch.com >> > > >> > > > https://freeswitch.com >> > > > >> > > > Official FreeSWITCH Sites >> > > > https://freeswitch.com/oss >> > > > https://freeswitch.org/confluence >> > > > https://cluecon.com >> > > > >> > > > FreeSWITCH-users mailing list >> > > > FreeSWITCH-users at lists.freeswitch.org >> > >> > > > > > >> > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > > >> > > >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > > > https://freeswitch.com >> > > >> > > >> > > >> > > >> > > ---------- Forwarded message ---------- >> > > From: Tamer Higazi via FreeSWITCH-users >> > > > > >> > > > > >> >> > > To: freeswitch-users at lists.freeswitch.org >> > >> > > > > > >> > > Cc: >> > > Bcc: >> > > Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) >> > > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > > permissions issue >> > > >> > >> _________________________________________________________________________ >> > > >> > > 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 >> > >> > >> > >> > >> > ---------- Forwarded message ---------- >> > From: Tamer Higazi via FreeSWITCH-users >> > > > > >> > To: freeswitch-users at lists.freeswitch.org >> > >> > Cc: >> > Bcc: >> > Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) >> > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio >> > permissions issue >> > >> _________________________________________________________________________ >> > >> > 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 >> >> >> >> >> ---------- Forwarded message ---------- >> From: Tamer Higazi via FreeSWITCH-users < >> freeswitch-users at lists.freeswitch.org> >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Bcc: >> Date: Sun, 02 Feb 2020 19:58:09 -0800 (PST) >> Subject: Re: [Freeswitch-users] Stumped by mod_portaudio permissions issue >> _________________________________________________________________________ >> >> 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 bipin at xbipin.com Sun Feb 16 09:49:55 2020 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 16 Feb 2020 13:49:55 +0400 Subject: [Freeswitch-users] SIP/RTP to TLS/SRTP calls Message-ID: <411b0c1d4a1f0940cf1d30b6be2d6c63@xbipin.com> hi, i have 2 profiles where one is purely SIP/RTP and the second is TLS/SRTP only, i have clients registering to both. If a TLS registered client calls a SIP based one everything works fine as FS uses SRTP for orginal leg and RTP for the outbound leg but if a SIP registered client calls a TLS one then the audio fails as FS tries to send RTP rather than SRTP. I have late negotiation and inherit codecs enabled. Is there any way to check in dialplan if the registered user is TLS based and use SRTP for that leg of the call and use normal SIP/RTP for the originating leg or some profile param which can force SRTP for all clients registered to the TLS profile and RTP for all clients registered to the SIP profile the way im currently getting around this is using the below but it isnt elegant then i bridge the call, if the above fails it means its a SIP to SIP call -- Regards, Bipin From sdevoy at bizfocused.com Sun Feb 16 18:56:45 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 16 Feb 2020 18:56:45 +0000 Subject: [Freeswitch-users] SQLite Warnings Message-ID: I am getting these SQLite warnings on a pretty new sever with 3 phones registered and NONE in use. [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] ... Why? It is a dedicated hardware server with i5 and 8 GB ram. Debian 10 and FS 1.10 Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: From borik.internet at gmail.com Sun Feb 16 19:34:11 2020 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Sun, 16 Feb 2020 22:34:11 +0300 Subject: [Freeswitch-users] SQLite Warnings In-Reply-To: References: Message-ID: Have the same problem. DB on a memory drive (/var/llib/freeswitch/db -> /dev/shm/fs_db). Near 25 CAPS/1000sessions in parallel. Debian Buster, 8 cores, 16GB memory, KVM/proxmox virtual. вс, 16 февр. 2020 г. в 22:23, Sean Devoy : > I am getting these SQLite warnings on a pretty new sever with 3 phones > registered and NONE in use. > > [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] … > > > > Why? It is a dedicated hardware server with i5 and 8 GB ram. Debian 10 > and FS 1.10 > > > > Thanks, > > Sean > _________________________________________________________________________ > > 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 andywolk at gmail.com Sun Feb 16 20:07:22 2020 From: andywolk at gmail.com (Andrey Wolk) Date: Mon, 17 Feb 2020 00:07:22 +0400 Subject: [Freeswitch-users] SQLite Warnings In-Reply-To: References: Message-ID: There are 300 tries to complete each request. SQLite is fast and most of the time completes requests right away, but sometimes it may be busy for a real bit. This exact line tells you that there are 299 tries left so it is OK. But if that number goes lower this may really help you measuring and understand the load. When it's 0 then that's a real problem and a request was not complete. Until that, never mind. Your machine needs to be very busy to go that low. вс, 16 февр. 2020 г. в 23:32, Sean Devoy : > I am getting these SQLite warnings on a pretty new sever with 3 phones > registered and NONE in use. > > [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] … > > > > Why? It is a dedicated hardware server with i5 and 8 GB ram. Debian 10 > and FS 1.10 > > > > Thanks, > > Sean > _________________________________________________________________________ > > 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 sdevoy at bizfocused.com Mon Feb 17 18:38:04 2020 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 17 Feb 2020 18:38:04 +0000 Subject: [Freeswitch-users] SQLite Warnings In-Reply-To: References: Message-ID: Thanks Andrew. I appreciate your explanation so much more than just “Don’t worry about”. It does mean “don’t worry about it”, lol, but so much more. Now I know what to watch for. Just curious, would MySQL be better? Sean From: FreeSWITCH-users On Behalf Of Andrey Wolk Sent: Sunday, February 16, 2020 3:07 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SQLite Warnings There are 300 tries to complete each request. SQLite is fast and most of the time completes requests right away, but sometimes it may be busy for a real bit. This exact line tells you that there are 299 tries left so it is OK. But if that number goes lower this may really help you measuring and understand the load. When it's 0 then that's a real problem and a request was not complete. Until that, never mind. Your machine needs to be very busy to go that low. вс, 16 февр. 2020 г. в 23:32, Sean Devoy >: I am getting these SQLite warnings on a pretty new sever with 3 phones registered and NONE in use. [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] … Why? It is a dedicated hardware server with i5 and 8 GB ram. Debian 10 and FS 1.10 Thanks, Sean _________________________________________________________________________ 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 mrjoli021 at gmail.com Mon Feb 17 20:37:53 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Mon, 17 Feb 2020 15:37:53 -0500 Subject: [Freeswitch-users] HA Cluster build Message-ID: Hello, We currently have two Freeswitch servers sharing a MariaDB. The plan is to have both server be an HA pair, so if we lose one server the other take over the active calls without dropping them. I have read online several ways of doing this, but most of them include using a floating IP. The problem with that is we would need to change our IP's and at this point we dont have that option. Is there a different approach to doing this? And if so could you send me an article or point me in the right direction? thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: From piotr at dataandsignal.com Mon Feb 17 22:41:38 2020 From: piotr at dataandsignal.com (Piotr Gregor) Date: Mon, 17 Feb 2020 22:41:38 +0000 Subject: [Freeswitch-users] JsSip Client receiving "488 Not Acceptable Here" when trying to make a call In-Reply-To: References: Message-ID: Hi Mahmood, Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip. var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, }; var session = ua.call(uri, options); kind regards, Piotr Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < mahmood.alkhalil at outlook.com> wrote: > Hi Everyone, > > I've been trying to use JsSIp over Websockets, I'm able to register and i > can confirm the registration is a success, but whenevr i try to make a call > Freeswitch responds with SIP/2.0 488 Not Acceptable Here. > > ​The JsSip API is showing SDP Failure, in Freeswitch logs it is telling > that there is SDP/RTP matches but then it say no candidate found. > > I'v attached the call trace, appreciate any help. > > Thanks, > Mahmood Alkhalil. > _________________________________________________________________________ > > 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 Feb 18 01:51:19 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 18 Feb 2020 01:51:19 +0000 Subject: [Freeswitch-users] JsSip Client receiving "488 Not Acceptable Here" when trying to make a call In-Reply-To: References: Message-ID: Get a trace and check your codecs. On Mon, 17 Feb 2020 at 22:57, Piotr Gregor wrote: > Hi Mahmood, > > > Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing > between ends of the call. > Define ICE servers to be used by JsSip. > > var options = { > 'eventHandlers' : your eventHandlers, > 'mediaConstraints' : { 'audio': true, 'video': true }, > 'pcConfig' : { 'iceServers': [ > {'urls': 'stun:stun.stunprotocol.org:3478'}, > {'urls': 'stun:stun.l.google.com:19302'}, ] }, > }; > > var session = ua.call(uri, options); > > > kind regards, > Piotr > > > Piotr Gregor > Software Engineer > > M: (+44) 07483 866 525 www: dataandsignal.com > > > > > > > On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil < > mahmood.alkhalil at outlook.com> wrote: > >> Hi Everyone, >> >> I've been trying to use JsSIp over Websockets, I'm able to register and i >> can confirm the registration is a success, but whenevr i try to make a call >> Freeswitch responds with SIP/2.0 488 Not Acceptable Here. >> >> ​The JsSip API is showing SDP Failure, in Freeswitch logs it is telling >> that there is SDP/RTP matches but then it say no candidate found. >> >> I'v attached the call trace, appreciate any help. >> >> Thanks, >> Mahmood Alkhalil. >> _________________________________________________________________________ >> >> 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 th982a at googlemail.com Tue Feb 18 03:11:38 2020 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 18 Feb 2020 04:11:38 +0100 Subject: [Freeswitch-users] Stumped by mod_portaudio permissions issue In-Reply-To: References: <20150826153850.GA6402@hau.nz> <20150901154401.GA7510@hau.nz> <20150910180739.GB5710@hau.nz> <12389.1441980578@ccs.covici.com> <20150911164506.GA8640@hau.nz> Message-ID: Hi Tom, Have you looked in this unit file ? [Service] User=tamer Group=users Type=forking PIDFile=/opt/freeswitch/var/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/opt/freeswitch/bin/freeswitch -nc replace User with YOUR username and GROUP with the usernames standard GROUP and if that does not work, replace: ExecStart=/opt/freeswitch/bin/freeswitch -nc with ExecStart=/opt/freeswitch/bin/freeswitch -nc -u tamer -g users where tamer is your username and users is the usergroup your useraccount runs. best, Tamer On 2020-02-16 06:58, Tom Lynn wrote: > When I run your example unit file, I found that yes, the pa devlist > command works, but freeswitch isn't running as the user.  it is > running as root, which is what I'm forbidden from doing. > > On Sun, Feb 2, 2020 at 9:46 PM Tom Lynn > wrote: > > Thank you, Tamer! > > I'll try your unit file sometime this week.   > > On Sun, Feb 2, 2020 at 7:58 PM Tamer Higazi via FreeSWITCH-users > > wrote: > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi > > To: freeswitch-users at lists.freeswitch.org > > Cc:  > Bcc:  > Date: Mon, 3 Feb 2020 04:57:26 +0100 > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > Dear Tom, > > It doesn't matter. It works, it shows you all devices. > > If you can run as usual user "alsamixer" and be capable to > list the > soundcard and you run freeswitch where all files are owned by > this user, > and start freeswitch as well as this user then you can access > your > soundcard with mod_portaudio without any problems. > > here is my unit file: > > [Unit] > Description=freeswitch > After=syslog.target network.target local-fs.target > > [Service] > User=tamer > Group=users > Type=forking > PIDFile=/opt/freeswitch/var/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/opt/freeswitch/bin/freeswitch -nc > TimeoutSec=45s > Restart=on-failure > ; exec > WorkingDirectory=/opt/freeswitch/bin > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > WantedBy=multi-user.target > > > here: > > tamer at tux ~ $ fs_cli -r > freeswitch at tux> pa dev > > [             devlist] > > > freeswitch at tux> pa devlist > 0;HDA NVidia: HDMI 0 (hw:0,3)(ALSA);0;8; > 1;HDA NVidia: HDMI 1 (hw:0,7)(ALSA);0;8; > 2;HDA NVidia: HDMI 2 (hw:0,8)(ALSA);0;2; > 3;HDA NVidia: HDMI 3 (hw:0,9)(ALSA);0;8; > 4;HD-Audio Generic: ALC892 Analog (hw:1,0)(ALSA);2;2; > 5;Logitech Webcam C925e: USB Audio (hw:2,0)(ALSA);2;0; > 6;Sound Blaster E5: USB Audio (hw:3,0)(ALSA);2;2; > 7;sysdefault(ALSA);128;128; > 8;front(ALSA);0;2; > 9;surround40(ALSA);0;2; > 10;iec958(ALSA);0;2; > 11;spdif(ALSA);2;2; > 12;pulse(ALSA);32;32;r,i,o > 13;upmix(ALSA);8;8; > 14;vdownmix(ALSA);6;6; > 15;dmix(ALSA);0;2; > 16;default(ALSA);32;32; > > freeswitch at tux> > > best, Tamer > > On 2020-02-03 01:43, Tom Lynn wrote: > > One more item, I still don't have pulseaudio installed and > the pa > > commands work, so pulseaudio is not strictly necessary. > > > > On Sun, Feb 2, 2020 at 4:33 PM Tom Lynn > > >> wrote: > > > >     I followed your steps, and yes, pa commands function when > >     freeswitch is run from the command line.  Nice! > > > >     I tried running it as both root and as user tom.  Both > times I was > >     able to use the pa commands with success.  I tried > running user > >     tom group user from the systemd startup and was not > successful > >     using pa commands. > > > >     The question becomes, why does this not work using the > >     freeswitch.service configuration file supplied with the > packaged > >     freeswitch?  I installed freeswitch-meta-all. > > > >     Tom > > > >     On Sun, Feb 2, 2020 at 9:02 AM Tamer Higazi via > FreeSWITCH-users > >      > >      >> wrote: > > > > > > > > > >         ---------- Forwarded message ---------- > >         From: Tamer Higazi > >          >> > >         To: freeswitch-users at lists.freeswitch.org > > >          > > >         Cc: > >         Bcc: > >         Date: Sun, 2 Feb 2020 18:02:09 +0100 > >         Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > >         permissions issue > >         Dear Tom, > > > >         This is not true. > >         I am running freeswitch as the user I am logged in. > > > >         as root: > >         chown -R tamer:users /opt/freeswitch > > > >         as user: > >         /opt/freeswitch/bin/freeswitch -nc > > > >         as user: > >         fs_cli -r > > > >         and you will see portaudio runs without any problems. > > > >         I have it in combination with pulseaudio running. > > > > > >         best, Tamer > > > >         On 2020-02-02 17:01, Tom Lynn wrote: > >         > I do not have pulse audio installed. > >         > > >         > I can access alsamixer and aplay from the shell > while logged > >         in as > >         > either root or freeswitch without issues. > >         > > >         > mod_portaudio only works WHEN I'm running > freeswitch under > >         group root, > >         > which I will not be able to do in a production > environment. > >         > > >         > On Sun, Feb 2, 2020 at 7:24 AM Tamer Higazi via > >         FreeSWITCH-users > >         > > >          > > >         > > >          >>> wrote: > >         > > >         > > >         > > >         > > >         >     ---------- Forwarded message ---------- > >         >     From: Tamer Higazi > >          > > >         >      > >          >>> > >         >     To: freeswitch-users at lists.freeswitch.org > > >          > > >         >      > >          >> > >         >     Cc: > >         >     Bcc: > >         >     Date: Sun, 2 Feb 2020 15:22:31 +0100 > >         >     Subject: Re: [Freeswitch-users] Stumped by > mod_portaudio > >         >     permissions issue > >         >     Dear Mark, > >         > > >         >     I tell you exactly why. > >         > > >         >     That is less a permission issue. > >         > > >         >     if you open the shell as "root" and type > alsamixer you > >         won't be > >         >     capable > >         >     to access the soundcard because "root" is not > in the > >         1st) sound group > >         >     and 2nd) root ha no access to the pulseaudio > server > >         > > >         >     This is why I MUST run freeswitch as the user > that is > >         currently > >         >     logged in. > >         > > >         >     mod_alsa doesn't work at all. I didn't get it > running, only > >         >     mod_portaudio.... > >         > > >         > > >         >     best, Tamer > >         > > >         > > >         >     On 2020-02-02 07:59, Tom Lynn wrote: > >         >     > For what it's worth, I'm running into this > very same > >         issue over 4 > >         >     > years after the original poster.  I suspect that > >         leaving off the > >         >     group > >         >     > option is leaving the process running as > group root, > >         which would > >         >     mimic > >         >     > my attempt at having the service run as user > >         freeswitch/group root, > >         >     > which allows mod_portaudio to function. > >         >     > > >         >     > I've posted here in the mail list, on IRC and in > >         Slack, but the > >         >     only > >         >     > bite was someone hinting about freeswitch > user needing > >         to be in > >         >     group > >         >     > audio, which it is.  This is broken. I've > looked at > >         submitting a > >         >     JIRA > >         >     > on this, but my login doesn't appear to > allow it. > >         >     > > >         >     > On Fri, Sep 11, 2015 at 9:45 AM Mark Haun > >          > > > >         >      >> > >         >     > > > >          > >>>> wrote: > >         >     > > >         >     > covici at ccs.covici.com > > > >          >> > >         >      > >          > > >          >>> > >         >     >     [covici at ccs.covici.com > > >          > > >          >> > >         >      > >          > > >          >>>] wrote: > >         >     >     > If  using alsa, did you check the > permission of > >         /dev/snd > >         >     and its > >         >     >     > children? > >         >     > > >         >     >     Yes.  In fact, both alsa operations > (aplay) and > >         portaudio > >         >     >     enumeration work > >         >     >     correctly when running as the freeswitch > user.  > >         All of this is > >         >     >     detailed in > >         >     >     my initial post, two or three weeks ago. > >         >     > > >         >     >     Mark > >         >     > > >         >     > > >         > > >        >    _________________________________________________________________________ > >         >     >     Professional FreeSWITCH Consulting Services: > >         >     > consulting at freeswitch.org > > >          > > >          > >          >> > >         >      > >          > > >          > >          >>> > >         >     > http://www.freeswitchsolutions.com > >         >     > > >         >     >     Official FreeSWITCH Sites > >         >     > http://www.freeswitch.org > >         >     > http://confluence.freeswitch.org > >         >     > http://www.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 > >         >     > http://www.freeswitch.org > >         >     > > >         >     > > >         >     > > >         > > >        >   _________________________________________________________________________ > >         >     > > >         >     > The FreeSWITCH project is sponsored by > SignalWire > >         > https://signalwire.com > >         >     > Enhance your FreeSWITCH install with > disruptive priced > >         SMS and > >         >     PSTN services. > >         >     > Build your next product on our scalable > cloud platform. > >         >     > > >         >     > Join our online community to chat in real time > >         > https://signalwire.community > >         >     > > >         >     > Professional FreeSWITCH Services > >         >     > sales at freeswitch.com > > > >          >> > >         >     > https://freeswitch.com > >         >     > > >         >     > Official FreeSWITCH Sites > >         >     > https://freeswitch.com/oss > >         >     > https://freeswitch.org/confluence > >         >     > https://cluecon.com > >         >     > > >         >     > FreeSWITCH-users mailing list > >         >     > FreeSWITCH-users at lists.freeswitch.org > > >          > > >         >      > >          >> > >         >     > > >        >  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >         >     > > >         >    > >        >   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >         >     > https://freeswitch.com > >         > > >         > > >         > > >         > > >         >     ---------- Forwarded message ---------- > >         >     From: Tamer Higazi via FreeSWITCH-users > >         >      > >          > > >         >      > >          >>> > >         >     To: freeswitch-users at lists.freeswitch.org > > >          > > >         >      > >          >> > >         >     Cc: > >         >     Bcc: > >         >     Date: Sun, 02 Feb 2020 07:24:11 -0800 (PST) > >         >     Subject: Re: [Freeswitch-users] Stumped by > mod_portaudio > >         >     permissions issue > >         > > >        >   _________________________________________________________________________ > >         > > >         >     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 > > > > > > > > > >         ---------- Forwarded message ---------- > >         From: Tamer Higazi via FreeSWITCH-users > >          > >          >> > >         To: freeswitch-users at lists.freeswitch.org > > >          > > >         Cc: > >         Bcc: > >         Date: Sun, 02 Feb 2020 09:02:50 -0800 (PST) > >         Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > >         permissions issue > >        >  _________________________________________________________________________ > > > >         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 > > > > > ---------- Forwarded message ---------- > From: Tamer Higazi via FreeSWITCH-users > > > To: freeswitch-users at lists.freeswitch.org > > Cc:  > Bcc:  > Date: Sun, 02 Feb 2020 19:58:09 -0800 (PST) > Subject: Re: [Freeswitch-users] Stumped by mod_portaudio > permissions issue > _________________________________________________________________________ > > 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 gmaruzz at gmail.com Tue Feb 18 05:40:50 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 18 Feb 2020 06:40:50 +0100 Subject: [Freeswitch-users] HA Cluster build In-Reply-To: References: Message-ID: No, there is no way but moving a floating ip adress (you can use some black magic bgp-fu, but is probably outside your reach). Also, your topology (two FS in HA) is the most difficult and less efficient. You want to use a couple proxy (opensips, kamailio) in HA, that distribute calls to N freeswitches. This way you achieve robust HA, load balancing, and all FS machines (heavy duty) are active, while only one proxy machine (very light) is passive. On Mon, Feb 17, 2020, 21:38 Joli Martinez wrote: > Hello, > > We currently have two Freeswitch servers sharing a MariaDB. > > The plan is to have both server be an HA pair, so if we lose one server > the other take over the active calls without dropping them. > > I have read online several ways of doing this, but most of them include > using a floating IP. The problem with that is we would need to change our > IP's and at this point we dont have that option. > > Is there a different approach to doing this? And if so could you send me > an article or point me in the right direction? > > thanks, > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From jalsot at gmail.com Tue Feb 18 09:52:42 2020 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Tue, 18 Feb 2020 10:52:42 +0100 Subject: [Freeswitch-users] HA Cluster build In-Reply-To: References: Message-ID: This might be of interest to you (I haven't tried): https://github.com/signalwire/freeswitch/pull/322 On Tue, 18 Feb 2020 at 07:04, Giovanni Maruzzelli wrote: > No, there is no way but moving a floating ip adress (you can use some > black magic bgp-fu, but is probably outside your reach). > > Also, your topology (two FS in HA) is the most difficult and less > efficient. > > You want to use a couple proxy (opensips, kamailio) in HA, that distribute > calls to N freeswitches. This way you achieve robust HA, load balancing, > and all FS machines (heavy duty) are active, while only one proxy machine > (very light) is passive. > > > > On Mon, Feb 17, 2020, 21:38 Joli Martinez wrote: > >> Hello, >> >> We currently have two Freeswitch servers sharing a MariaDB. >> >> The plan is to have both server be an HA pair, so if we lose one server >> the other take over the active calls without dropping them. >> >> I have read online several ways of doing this, but most of them include >> using a floating IP. The problem with that is we would need to change our >> IP's and at this point we dont have that option. >> >> Is there a different approach to doing this? And if so could you send me >> an article or point me in the right direction? >> >> 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 > > _________________________________________________________________________ > > 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 nneul at mst.edu Tue Feb 18 12:02:03 2020 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 18 Feb 2020 06:02:03 -0600 Subject: [Freeswitch-users] HA Cluster build In-Reply-To: References: Message-ID: <26cdf6df-c0c7-5e80-0726-8df2b90137eb@mst.edu> An HTML attachment was scrubbed... URL: From mrjoli021 at gmail.com Thu Feb 20 16:44:01 2020 From: mrjoli021 at gmail.com (Joli Martinez) Date: Thu, 20 Feb 2020 11:44:01 -0500 Subject: [Freeswitch-users] ha-error messages Message-ID: Hello, I have two freeswitch 1.6 boxes running on Debian. I created an external DB server and both servers are connected to the same external MariaDB. I also added a floating IP to both boxes. All calls and registrations are coming into the floating IP. The A side is the Master. When run fs_cli on both boxes I constantly get the following errors on both boxes. Not sure how to fix them. A Side ############## 2020-02-20 10:34:01.889335 [ERR] sofia_reg.c:2447 1112223333-GW2 Failed Registration with status Request Timeout [408]. failure #11 2020-02-20 10:34:01.889335 [ERR] sofia_reg.c:2447 1112223333-GW1 Failed Registration with status Request Timeout [408]. failure #11 2020-02-20 10:34:01.889335 [ERR] sofia_reg.c:2447 1112224444-GW2 Failed Registration with status Request Timeout [408]. failure #11 2020-02-20 10:34:01.889335 [ERR] sofia_reg.c:2447 1112224444-GW1 Failed Registration with status Request Timeout [408]. failure #11 B Side ############## 2020-02-20 10:34:31.383548 [ERR] switch_odbc.c:522 ERR: [update channels set application='log',application_data='INFO \',presence_id=' 1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='23e02070-23e0-4242-8175-5aa7cdeb2eea'] [STATE: 42000 CODE 1064 ERROR: [ma-2.0.13][5.5.64-MariaDB]You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='' at line 1 ] 2020-02-20 10:34:31.383548 [ERR] switch_core_sqldb.c:587 ODBC SQL ERR [STATE: 42000 CODE 1064 ERROR: [ma-2.0.13][5.5.64-MariaDB]You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='' at line 1 ] update channels set application='log',application_data='INFO \',presence_id='1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='23e02070-23e0-4242-8175-5aa7cdeb2eea' 2020-02-20 10:34:31.623593 [ERR] switch_odbc.c:522 ERR: [update channels set application='log',application_data='INFO \',presence_id=' 1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='23e02070-23e0-4242-8175-5aa7cdeb2eea'] [STATE: 42000 CODE 1064 ERROR: [ma-2.0.13][5.5.64-MariaDB]You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='' at line 1 ] 2020-02-20 10:34:31.623593 [ERR] switch_core_sqldb.c:587 ODBC SQL ERR [STATE: 42000 CODE 1064 ERROR: [ma-2.0.13][5.5.64-MariaDB]You have an error in your SQL syntax; check the manual that corresponds to your MariaDB server version for the right syntax to use near '1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='' at line 1 ] update channels set application='log',application_data='INFO \',presence_id='1112223333 at sip.host.com',presence_data='',accountcode='' where uuid='23e02070-23e0-4242-8175-5aa7cdeb2eea' -------------- next part -------------- An HTML attachment was scrubbed... URL: From gd at medat.de Thu Feb 20 15:09:13 2020 From: gd at medat.de (Gert Doering) Date: Thu, 20 Feb 2020 16:09:13 +0100 Subject: [Freeswitch-users] event socket, mod_spandsp, remote tif files? Message-ID: <20200220150913.GU40894704@medat.de> Good morning, I am working on glueing mgetty+sendfax and freeswitch/spandsp together (sort of "like GoFaxIP", just not hylafax, can't help myself here). Using the freeswitch event socket is easy enough, but so far requires me to copy over the .tif file to the freeswitch server, to reference the file name for spandsp to send out. The documentation (I could find so far) mentions the "unicast" command in the context of mod_spandsp, but I think this is just for UDP RTP data (so mod_spandsp could run remotely), and I have not found a way to transfer "fax files" with it. So, question to the group - if you have a freeswitch server with mod_spandsp, and you have a remote host with an event socket client and a .tif file to be sent, is there a way to "pass over" that .tif file without copying it onto the server's file system first? (CIFS, NFS, etc. all fall under "copy onto the server's file system" in a way) - the idea is to have the server fairly much locked down, and only talk to it from the fax client via event socket. thanks, gert -- MEDAT Computer-Systeme GmbH +--------------------------------------------- Albrecht. 14 | HRB Muenchen 56 206, USt-IdNr. DE129411894 D-80636 Muenchen | Geschaeftsfuehrer: Tel: +49 89 126808-0 | - Stefan Henkelmann Fax: +49 89 126808-50 | M: g.doering at medat.de + From gd at medat.de Thu Feb 20 15:09:13 2020 From: gd at medat.de (Gert Doering) Date: Thu, 20 Feb 2020 16:09:13 +0100 Subject: [Freeswitch-users] event socket, mod_spandsp, remote tif files? Message-ID: <20200220150913.GU40894704@medat.de> Good morning, I am working on glueing mgetty+sendfax and freeswitch/spandsp together (sort of "like GoFaxIP", just not hylafax, can't help myself here). Using the freeswitch event socket is easy enough, but so far requires me to copy over the .tif file to the freeswitch server, to reference the file name for spandsp to send out. The documentation (I could find so far) mentions the "unicast" command in the context of mod_spandsp, but I think this is just for UDP RTP data (so mod_spandsp could run remotely), and I have not found a way to transfer "fax files" with it. So, question to the group - if you have a freeswitch server with mod_spandsp, and you have a remote host with an event socket client and a .tif file to be sent, is there a way to "pass over" that .tif file without copying it onto the server's file system first? (CIFS, NFS, etc. all fall under "copy onto the server's file system" in a way) - the idea is to have the server fairly much locked down, and only talk to it from the fax client via event socket. thanks, gert -- MEDAT Computer-Systeme GmbH +--------------------------------------------- Albrecht. 14 | HRB Muenchen 56 206, USt-IdNr. DE129411894 D-80636 Muenchen | Geschaeftsfuehrer: Tel: +49 89 126808-0 | - Stefan Henkelmann Fax: +49 89 126808-50 | M: g.doering at medat.de + From iriik at yahoo.com Sat Feb 15 22:11:12 2020 From: iriik at yahoo.com (I K) Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) Subject: RTP not transmitting over NAT References: <622480823.3536006.1581804672072.ref@mail.yahoo.com> Message-ID: <622480823.3536006.1581804672072@mail.yahoo.com> Hi every one, I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?! Things I've acheived so far:1- bypass-media = false and proxy-media = false:In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well. 2- bypass-media = false and proxy-media = true:audio works but video shows blank3- bypass-media = true and proxy-media = false:If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s. I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jason.holden at discountpbx.com Wed Feb 19 17:39:46 2020 From: jason.holden at discountpbx.com (Jason Holden) Date: Wed, 19 Feb 2020 12:39:46 -0500 Subject: [Freeswitch-users] shared line CID problem Message-ID: <000001d5e74b$9486e6d0$bd94b470$@discountpbx.com> Hi, Currently experiencing a problem with SLA on Polycoms where when the call is answered the CID name / number changes to be the phoens extension number and no longer shows the incoming cid of the caller. This is also experienced when eextension to extension dialing but is more speratic where 20% of the time after the bridge is answered the CID changes but the other 80% of the time for extension dialing its fine. I have the same problem on FS versions 1.8.56 1.8.7 and the latest 1.10. Does anyone have any recommendations? -------------- next part -------------- An HTML attachment was scrubbed... URL: From MomoSxp at hotmail.com Thu Feb 20 19:48:11 2020 From: MomoSxp at hotmail.com (MomoSxp at hotmail.com) Date: Thu, 20 Feb 2020 19:48:11 +0000 Subject: [Freeswitch-users] how to close conference after playing audio files Message-ID: im trying to start a conference between a list of users, play a list of audio files and then close the conference. im using the event_socket_library in c++ and am doing the following: conference dial user/user1 conference dial user/user2 conference dial user/user3 in a loop: conference play conference hup all //will skip the stream and end the call, but i need to wait for the last playback to end. how do i know when the last file was played so i can close the conference? is there a way to do it programmatically? a dialplan script taking the users and the files and the conference name as parameters, starting and closing the conference automatically would be fine, too. doing this im not sure on how to play a file to a conference and closing it. i tried something like this: but its not playing the file and not closing the conference -------------- next part -------------- An HTML attachment was scrubbed... URL: From moshe.rosenberg at gmail.com Sun Feb 16 20:13:00 2020 From: moshe.rosenberg at gmail.com (Moshe Rosenberg) Date: Sun, 16 Feb 2020 15:13:00 -0500 Subject: [Freeswitch-users] SQLite Warnings In-Reply-To: References: Message-ID: This is something that FS used not to show in the older version in the logs This was add it at version 1.10.2 I believe this error was always there but was not displayed There is nothing to worry about. All it does FS will try again .. if you see the same line twice it might be an issue On Sun, Feb 16, 2020, 3:02 PM Dmitriy Borisov wrote: > Have the same problem. DB on a memory drive (/var/llib/freeswitch/db -> > /dev/shm/fs_db). Near 25 CAPS/1000sessions in parallel. Debian Buster, 8 > cores, 16GB memory, KVM/proxmox virtual. > > вс, 16 февр. 2020 г. в 22:23, Sean Devoy : > >> I am getting these SQLite warnings on a pretty new sever with 3 phones >> registered and NONE in use. >> >> [WARNING] switch_core_db.c:92 SQLite is BUSY, sane=299 [BEGIN EXCLUSIVE] … >> >> >> >> Why? It is a dedicated hardware server with i5 and 8 GB ram. Debian 10 >> and FS 1.10 >> >> >> >> Thanks, >> >> Sean >> _________________________________________________________________________ >> >> 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 imfanee at gmail.com Fri Feb 21 03:52:52 2020 From: imfanee at gmail.com (Faisal Hanif) Date: Fri, 21 Feb 2020 08:52:52 +0500 Subject: [Freeswitch-users] RTP not transmitting over NAT In-Reply-To: References: <622480823.3536006.1581804672072.ref@mail.yahoo.com> Message-ID: In you are using linphone mobile clients make sure you have added " stun.linphone.org" in account specific stun settings, If you need both ZRTP & NAT handling keep media in proxy mode. On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: I K > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Cc: > Bcc: > Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) > Subject: RTP not transmitting over NAT > Hi every one, > > I'm new to Freeswitch and I'm having some difficulty configuring it, > despite reading the documents. I have a Freeswitch running on a server with > public IP and several Linphone android clients running behind NAT. > How should I configure it to transfer audio and video and use ZRTP if one > client is on a LAN behind NAT and the other client is using mobile data?! > > Things I've acheived so far: > 1- bypass-media = false and proxy-media = false: > In this mode audio works but when I turn on video no request is sent to > the other phone. It other use just has to know when to turn on the camera > to receive video. and of course ZRTP does not work well. > 2- bypass-media = false and proxy-media = true: > audio works but video shows blank > 3- bypass-media = true and proxy-media = false: > If both clients are on the same LAN everything works fine, but if one > client connects via mobile data, they can still call each other but no > media is transferred and the call is dropped after 30s. > > I've been working on this scenario for over a week and I'm desperate to > make it work. Thanks in advance for your help. > > > > > > ---------- Forwarded message ---------- > From: I K via FreeSWITCH-users > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Cc: > Bcc: > Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) > Subject: [Freeswitch-users] RTP not transmitting over NAT > _________________________________________________________________________ > > 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 Fri Feb 21 07:57:52 2020 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 21 Feb 2020 10:57:52 +0300 Subject: [Freeswitch-users] how to close conference after playing audio files In-Reply-To: References: Message-ID: Think event "event_socket_library" able to catch "file_played" event. Just execute "/events all" in fs_cli and looks event generated after file played. Then try catch this using "event_socket_library". On Fri, Feb 21, 2020 at 5:11 AM MomoSxp at hotmail.com wrote: > im trying to start a conference between a list of users, play a list of > audio files and then close the conference. im using the > event_socket_library in c++ and am doing the following: > > conference dial user/user1 > conference dial user/user2 > conference dial user/user3 > in a loop: > conference play > conference hup all //will skip the stream and end the call, but i need to wait for the last playback to end. > > how do i know when the last file was played so i can close the conference? > is there a way to do it programmatically? a dialplan script taking the > users and the files and the conference name as parameters, starting and > closing the conference automatically would be fine, too. doing this im not > sure on how to play a file to a conference and closing it. > > i tried something like this: but its not playing the file and not closing > the conference > > > > > > > > > > > _________________________________________________________________________ > > 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 Fri Feb 21 08:38:19 2020 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 21 Feb 2020 11:38:19 +0300 Subject: [Freeswitch-users] RTP not transmitting over NAT In-Reply-To: References: <622480823.3536006.1581804672072.ref@mail.yahoo.com> Message-ID: Is without ZRTP video/addition removing works well? Case 1 Without ZRTP similar case fixed about two years ago. Sorry I cannot find related ticked created by me (jira login safarov). Please retest master and try bisect with commits 1-3 years ago. Think you will find working version and where bug is introduced. Case 2 Please look https://freeswitch.org/jira/browse/FS-9203 https://freeswitch.org/jira/browse/FS-9206 Here you will find commit where it worked properly. Just need bisect with master and find commit introduced bug. Case 3 Useless for you. This not designed for users behind NAT. So simple do not use. Sergey On Fri, Feb 21, 2020 at 7:22 AM Faisal Hanif wrote: > In you are using linphone mobile clients make sure you have added " > stun.linphone.org" in account specific stun settings, > If you need both ZRTP & NAT handling keep media in proxy mode. > > On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: I K >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) >> Subject: RTP not transmitting over NAT >> Hi every one, >> >> I'm new to Freeswitch and I'm having some difficulty configuring it, >> despite reading the documents. I have a Freeswitch running on a server with >> public IP and several Linphone android clients running behind NAT. >> How should I configure it to transfer audio and video and use ZRTP if one >> client is on a LAN behind NAT and the other client is using mobile data?! >> >> Things I've acheived so far: >> 1- bypass-media = false and proxy-media = false: >> In this mode audio works but when I turn on video no request is sent to >> the other phone. It other use just has to know when to turn on the camera >> to receive video. and of course ZRTP does not work well. >> 2- bypass-media = false and proxy-media = true: >> audio works but video shows blank >> 3- bypass-media = true and proxy-media = false: >> If both clients are on the same LAN everything works fine, but if one >> client connects via mobile data, they can still call each other but no >> media is transferred and the call is dropped after 30s. >> >> I've been working on this scenario for over a week and I'm desperate to >> make it work. Thanks in advance for your help. >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: I K via FreeSWITCH-users >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) >> Subject: [Freeswitch-users] RTP not transmitting over NAT >> _________________________________________________________________________ >> >> 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 Fri Feb 21 09:03:06 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 21 Feb 2020 09:03:06 +0000 Subject: [Freeswitch-users] RTP not transmitting over NAT In-Reply-To: References: <622480823.3536006.1581804672072.ref@mail.yahoo.com> Message-ID: If your fs is behind nat, you should set sip-ext-ip and rtp-ext-ip. This is done so that when fs sends an offer out it puts the sip-ext-ip and rtp ip in that offer. On the client side set the stun On Fri, 21 Feb 2020 at 04:38, Faisal Hanif wrote: > In you are using linphone mobile clients make sure you have added " > stun.linphone.org" in account specific stun settings, > If you need both ZRTP & NAT handling keep media in proxy mode. > > On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users < > freeswitch-users at lists.freeswitch.org> wrote: > >> >> >> >> ---------- Forwarded message ---------- >> From: I K >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) >> Subject: RTP not transmitting over NAT >> Hi every one, >> >> I'm new to Freeswitch and I'm having some difficulty configuring it, >> despite reading the documents. I have a Freeswitch running on a server with >> public IP and several Linphone android clients running behind NAT. >> How should I configure it to transfer audio and video and use ZRTP if one >> client is on a LAN behind NAT and the other client is using mobile data?! >> >> Things I've acheived so far: >> 1- bypass-media = false and proxy-media = false: >> In this mode audio works but when I turn on video no request is sent to >> the other phone. It other use just has to know when to turn on the camera >> to receive video. and of course ZRTP does not work well. >> 2- bypass-media = false and proxy-media = true: >> audio works but video shows blank >> 3- bypass-media = true and proxy-media = false: >> If both clients are on the same LAN everything works fine, but if one >> client connects via mobile data, they can still call each other but no >> media is transferred and the call is dropped after 30s. >> >> I've been working on this scenario for over a week and I'm desperate to >> make it work. Thanks in advance for your help. >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: I K via FreeSWITCH-users >> To: "freeswitch-users at lists.freeswitch.org" < >> freeswitch-users at lists.freeswitch.org> >> Cc: >> Bcc: >> Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) >> Subject: [Freeswitch-users] RTP not transmitting over NAT >> _________________________________________________________________________ >> >> 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 josen.figueroa at unixmexico.org Fri Feb 28 01:15:36 2020 From: josen.figueroa at unixmexico.org (Jose Figueroa) Date: Thu, 27 Feb 2020 19:15:36 -0600 Subject: [Freeswitch-users] How to get the call uuid from FS by verto client Message-ID: Hello folks, I'm trying to get the generated Freeswitch call uuid when I make a call using verto. Afaik I can export the variables from FS dialplan to verto clients using verto_h_* prefix and it works in all other scenarios, only not in this. I added this in the outbound calls dialplan but I cannot get those variables, such as uuid, country_name (I have a lua script to determine what country I'm dialing to), etc. I might be doing something wrong because according to this link https://stackoverflow.com/questions/52615360/how-do-i-pass-variables-from-freeswitch-dialplan-to-verto-client it says you could export variables to the Leg A or Leg B channels. I'm trying but I don't find the solution clearly. If someone managed this before, please let me know! it will be very appreciated. Thanks! Jose Figueroa -------------- next part -------------- An HTML attachment was scrubbed... URL: From atuxnull at gmail.com Fri Feb 21 12:40:38 2020 From: atuxnull at gmail.com (John Tuxies) Date: Fri, 21 Feb 2020 14:40:38 +0200 Subject: [Freeswitch-users] Freeswitch 32bits Message-ID: Trying to install freeswitch in a debian 10 32bits (ALIX 2d13) machine from source and i am getting the following error: *root at alix2d13: ~ $ apt-get build-dep freeswitchReading package lists... DoneReading package lists... DoneBuilding dependency treeReading state information... DoneSome packages could not be installed. This may mean that you haverequested an impossible situation or if you are using the unstabledistribution that some required packages have not yet been createdor been moved out of Incoming.The following information may help to resolve the situation:The following packages have unmet dependencies: builddeps:freeswitch : Depends: libks but it is not installable Depends: signalwire-client-c but it is not installable Depends: libbroadvoice-dev but it is not installable Depends: libsilk-dev but it is not installable Depends: libv8-6.1-dev but it is not installableE: Unable to correct problems, you have held broken packages.root at alix2d13: ~ $* No idea how to overcome this issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: From momosxp at hotmail.com Fri Feb 21 14:04:04 2020 From: momosxp at hotmail.com (momosxp) Date: Fri, 21 Feb 2020 07:04:04 -0700 (MST) Subject: [Freeswitch-users] how to close conference after playing audio files Message-ID: <1582293844116-0.post@n2.nabble.com> im trying to start a conference between a list of users, play a list of audio files and then close the conference. im using the event_socket_library in c++ and am doing the following: conference dial user/user1 conference dial user/user2 conference dial user/user3 in a loop: conference play conference hup all //will skip the stream and end the call, but i need to wait for the last playback to end. how do i know when the last file was played so i can close the conference? is there a way to do it programmatically? a dialplan script taking the users and the files and the conference name as parameters, starting and closing the conference automatically would be fine, too. doing this im not sure on how to play a file to a conference and closing it. i tried something like this: but its not playing the file and not closing the conference i tried looking for the event PLAYBACK_START and PLAYBACK_STOP, but they wont appear when using conferences. there are events for conferences like play-file and play-file-done: Table: Conference Events https://freeswitch.org/confluence/display/FREESWITCH/mod_conference But only the clients seem to receive the events. https://lists.freeswitch.org/pipermail/freeswitch-users/2016-August/121729.html Is there a way to make a dialplan for a conference that will close the conference on the condition that an audio file is finished playing? (the last one of multiple files) Or is there a way to receive the conference events with the event_socket_library? -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/ From atomjohnston at gmail.com Fri Feb 21 22:26:48 2020 From: atomjohnston at gmail.com (Adam Johnston) Date: Fri, 21 Feb 2020 17:26:48 -0500 Subject: [Freeswitch-users] play_and_get_digits on simultaneous calls Message-ID: I've been experimenting with IVRs and I'm having trouble with play_and_get_digits in mod_dptools. Both in XML and Lua I'm see play_and_get_digits process DTMF events from other calls that are simultaneously active. Below is an XML example of what I'm going for, just a basic authentication demo that reaches out to an HTTP endpoint via curl. If I try out the demo on simultaneous calls, key presses from one call can be picked up in the other and cause authentication to fail. Any ideas on how to isolate the DTMF capture to the individual sessions? -------------- next part -------------- An HTML attachment was scrubbed... URL: From victor.bogatyryev at gmail.com Sat Feb 22 07:37:06 2020 From: victor.bogatyryev at gmail.com (Victor Bogatyryev) Date: Sat, 22 Feb 2020 10:37:06 +0300 Subject: [Freeswitch-users] Sudden call termination: [terminating][487] Message-ID: <9bfe7cc6-db06-93de-4bd1-0926748f1f22@gmail.com> Hi, Sudden call termination about 30 s after the connection has been estableshed. There is no record in the SIPTRACE about sending BYE to/from the UA. tcpdump also does not see any SIP-related packets at the moment of call termination. FS hungup the channel but the UA continues to send rtp. tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 1372 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800e2e50 from (tls/33.2.151.64:2210) has 1372 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f30800a2f80): msg 0x7f30800e2e50 (1372 bytes) from tls/33.2.151.64:2210/sips next=(nil) nta.c:2880 agent_recv_request() nta: received INVITE sip:009197 at 6.27.9.120:5061 SIP/2.0 (CSeq 50) nta.c:3174 agent_check_request_via() nta: Via check: received=33.2.151.64 nta.c:3085 agent_recv_request() nta: INVITE (50) going to a default leg nta.c:1348 set_timeout() nta: timer shortened to 2000 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0x7f3080044490, 0x7f3080043c90, 0x7f3080063290) called soa.c:403 soa_set_params() soa_set_params(static::0x7f30800d65b0, ...) called nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7f30800f2c50) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f30800d65b0) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f30800d65b0, (nil), 0x7f30800eb52f, 637) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f3080063290): adding session usage nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_invite 100 Trying nua_session.c:4143 signal_call_state_change() nua(0x7f3080063290): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f30800d65b0, [0x7f30a14f97b8], [0x7f30a14f97c0], [(nil)]) called nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_state 100 Trying tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2020-02-22 08:56:56.867070 [NOTICE] switch_channel.c:1118 New Channel sofia/external/1000 at 6.27.9.120:5061 [223734d0-5538-11ea-bbfb-e979e915e05b] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:585 (sofia/external/1000 at 6.27.9.120:5061) Running State Change CS_NEW (Cur 1 Tot 9) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:56.867070 [DEBUG] sofia.c:10255 sofia/external/1000 at 6.27.9.120:5061 receiving invite from 33.2.151.64:2210 version: 1.10.2-release git f7bdd38 2019-12-31 14:01:19Z 64bit nua.c:610 nua_set_hparams() nua: nua_set_hparams: entering nua_stack.c:569 nua_stack_signal() nua(0x7f3080063290): recv signal r_set_params nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800d65b0, ...) called nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event r_set_params 200 OK nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0x7f3080063290): sent signal r_set_params nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:56.867070 [DEBUG] sofia.c:7301 Channel sofia/external/1000 at 6.27.9.120:5061 entering state [received][100] 2020-02-22 08:56:56.867070 [DEBUG] sofia.c:7311 Remote SDP: v=0 o=1000 8002 8000 IN IP4 192.168.2.5 s=SIP Call c=IN IP4 192.168.2.5 t=0 0 m=audio 5012 RTP/SAVP 8 0 4 18 97 102 2 103 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:102 G729E/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:103 AAL2-G726-40/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,32-36,54 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:djoxnaC5dQ7mre6tjhfK13Ca1sBgQKEIBjQab653|2^32 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:SCSxesFRMzZfGeNNxnFkkUjUKx6Ui182k2VqrtUY|2^32 2020-02-22 08:56:56.867070 [DEBUG] sofia.c:7714 (sofia/external/1000 at 6.27.9.120:5061) State Change CS_NEW -> CS_INIT nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:604 (sofia/external/1000 at 6.27.9.120:5061) State NEW 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:585 (sofia/external/1000 at 6.27.9.120:5061) Running State Change CS_INIT (Cur 1 Tot 9) 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:628 (sofia/external/1000 at 6.27.9.120:5061) State INIT 2020-02-22 08:56:56.867070 [DEBUG] mod_sofia.c:93 sofia/external/1000 at 6.27.9.120:5061 SOFIA INIT 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:40 sofia/external/1000 at 6.27.9.120:5061 Standard INIT 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:48 (sofia/external/1000 at 6.27.9.120:5061) State Change CS_INIT -> CS_ROUTING 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:628 (sofia/external/1000 at 6.27.9.120:5061) State INIT going to sleep 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:585 (sofia/external/1000 at 6.27.9.120:5061) Running State Change CS_ROUTING (Cur 1 Tot 9) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:2332 (sofia/external/1000 at 6.27.9.120:5061) Callstate Change DOWN -> RINGING 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:644 (sofia/external/1000 at 6.27.9.120:5061) State ROUTING nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:573 nua_stack_signal() nua(0x7f3080063290): recv signal r_respond 100 Trying nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800d65b0, ...) called tport.c:3273 tport_tsend() tport_tsend(0x7f30800a2f80) tpn = TLS/33.2.151.64:2210 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f30800a3420 0x7f30800ebe20 379 (379) tport.c:3610 tport_vsend() tport_vsend(0x7f30800a2f80): 379 bytes of 379 to tls/33.2.151.64:2210 tport.c:3508 tport_send_msg() tport_vsend returned 379 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:6797 incoming_reply() nta: sent 100 Trying for INVITE (50) nua_stack.c:529 nua_signal() nua(0x7f3080063290): sent signal r_respond 2020-02-22 08:56:56.867070 [DEBUG] mod_sofia.c:154 sofia/external/1000 at 6.27.9.120:5061 SOFIA ROUTING 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:236 sofia/external/1000 at 6.27.9.120:5061 Standard ROUTING 2020-02-22 08:56:56.867070 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->009197 in context public Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->unloop] continue=false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->outside_call] continue=true Dialplan: sofia/external/1000 at 6.27.9.120:5061 Absolute Condition [outside_call] Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action set(outside_call=true) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->call_debug] continue=true Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (FAIL) [public_extensions] destination_number(009197) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->public_conference_extensions] continue=false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (FAIL) [public_conference_extensions] destination_number(009197) =~ /^(3[5-8][01][0-9])$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->public_did] continue=false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (FAIL) [public_did] destination_number(009197) =~ /^(5551212)$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 parsing [public->us1_ipv6] continue=false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Regex (PASS) [us1_ipv6] destination_number(009197) =~ /^00(\d{3,})$/ break=on-false Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action export(inbound_late_negotiation=true) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action export(inbound_zrtp_passthru=false) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action set(rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action export(absolute_codec_string=PCMA) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action set(ringback=%(2000,4000,440,480)) Dialplan: sofia/external/1000 at 6.27.9.120:5061 Action bridge([rtp_secure_media=mandatory:AEAD_AES_256_GCM_8]sofia/gateway/us1_ipv6/9197) 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:287 (sofia/external/1000 at 6.27.9.120:5061) State Change CS_ROUTING -> CS_EXECUTE 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:644 (sofia/external/1000 at 6.27.9.120:5061) State ROUTING going to sleep 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:585 (sofia/external/1000 at 6.27.9.120:5061) Running State Change CS_EXECUTE (Cur 1 Tot 9) 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:651 (sofia/external/1000 at 6.27.9.120:5061) State EXECUTE 2020-02-22 08:56:56.867070 [DEBUG] mod_sofia.c:209 sofia/external/1000 at 6.27.9.120:5061 SOFIA EXECUTE 2020-02-22 08:56:56.867070 [DEBUG] switch_core_state_machine.c:329 sofia/external/1000 at 6.27.9.120:5061 Standard EXECUTE EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 set(outside_call=true) 2020-02-22 08:56:56.867070 [DEBUG] mod_dptools.c:1672 SET sofia/external/1000 at 6.27.9.120:5061 [outside_call]=[true] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 export(RFC2822_DATE=Sat, 22 Feb 2020 08:56:56 +0300) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 22 Feb 2020 08:56:56 +0300] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 export(inbound_late_negotiation=true) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [inbound_late_negotiation]=[true] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 export(inbound_zrtp_passthru=false) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [inbound_zrtp_passthru]=[false] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 set(rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_80) 2020-02-22 08:56:56.867070 [DEBUG] mod_dptools.c:1672 SET sofia/external/1000 at 6.27.9.120:5061 [rtp_secure_media]=[mandatory:AES_CM_128_HMAC_SHA1_80] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 export(absolute_codec_string=PCMA) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1310 EXPORT (export_vars) [absolute_codec_string]=[PCMA] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 set(ringback=%(2000,4000,440,480)) 2020-02-22 08:56:56.867070 [DEBUG] mod_dptools.c:1672 SET sofia/external/1000 at 6.27.9.120:5061 [ringback]=[%(2000,4000,440,480)] EXECUTE [depth=0] sofia/external/1000 at 6.27.9.120:5061 bridge([rtp_secure_media=mandatory:AEAD_AES_256_GCM_8]sofia/gateway/us1_ipv6/9197) 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1264 sofia/external/1000 at 6.27.9.120:5061 EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 22 Feb 2020 08:56:56 +0300] to event 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1264 sofia/external/1000 at 6.27.9.120:5061 EXPORTING[export_vars] [inbound_late_negotiation]=[true] to event 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1264 sofia/external/1000 at 6.27.9.120:5061 EXPORTING[export_vars] [inbound_zrtp_passthru]=[false] to event 2020-02-22 08:56:56.867070 [DEBUG] switch_channel.c:1264 sofia/external/1000 at 6.27.9.120:5061 EXPORTING[export_vars] [absolute_codec_string]=[PCMA] to event 2020-02-22 08:56:56.867070 [DEBUG] switch_ivr_originate.c:2212 Parsing global variables 2020-02-22 08:56:56.867070 [DEBUG] switch_ivr_originate.c:2760 Parsing session specific variables 2020-02-22 08:56:56.887055 [NOTICE] switch_channel.c:1118 New Channel sofia/external-ipv6/9197 [22386314-5538-11ea-bc07-e979e915e05b] 2020-02-22 08:56:56.887055 [DEBUG] mod_sofia.c:5089 (sofia/external-ipv6/9197) State Change CS_NEW -> CS_INIT 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:585 (sofia/external-ipv6/9197) Running State Change CS_INIT (Cur 2 Tot 10) 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:628 (sofia/external-ipv6/9197) State INIT 2020-02-22 08:56:56.887055 [DEBUG] mod_sofia.c:93 sofia/external-ipv6/9197 SOFIA INIT 2020-02-22 08:56:56.887055 [DEBUG] switch_core_media.c:1215 Set Local audio crypto Key [1 AEAD_AES_256_GCM_8 inline:l7g+SjnmyEXkdODYNoM4vjNljooxPJvFuumXJZM8OpgX52o5hnFWJ54qvhU=] 2020-02-22 08:56:56.887055 [DEBUG] switch_core_media.c:1215 Set Local video crypto Key [1 AEAD_AES_256_GCM_8 inline:UCkQ762LUOckln/vOdFBmuNp6gow4IM4WLQxgp1vMJ4XJMY3YLrmmt2KRlg=] 2020-02-22 08:56:56.887055 [DEBUG] switch_core_media.c:1215 Set Local text crypto Key [1 AEAD_AES_256_GCM_8 inline:b8+841hCfTOwh3wNzMustM9VWGHaCdw2kCm4rwH4LF9TLrpTUuLF8BhgnjI=] nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2020-02-22 08:56:56.887055 [DEBUG] sofia_glue.c:1618 sofia/external-ipv6/9197 sending invite version: 1.10.2-release git f7bdd38 2019-12-31 14:01:19Z 64bit Local SDP: v=0 o=FreeSWITCH 1582333804 1582333805 IN IP6 2a2a:810b:a2ba::65 s=FreeSWITCH c=IN IP6 2a2a:810b:a2ba::65 t=0 0 m=audio 17212 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AEAD_AES_256_GCM_8 inline:l7g+SjnmyEXkdODYNoM4vjNljooxPJvFuumXJZM8OpgX52o5hnFWJ54qvhU= a=ptime:20 a=sendrecv nua.c:633 nua_invite() nua: nua_invite: entering nua_stack.c:569 nua_stack_signal() nua(0x18ca7e0): recv signal r_invite nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0x7f307c0448f0, 0x7f307c0440f0, 0x18ca7e0) called soa.c:403 soa_set_params() soa_set_params(static::0x7f307c0b7770, ...) called soa.c:403 soa_set_params() soa_set_params(static::0x7f307c0b7770, ...) called soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7f307c0b7770, (nil), 0x17cec61, -1) called soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7f307c0b7770, (nil), 0x17cec61, -1) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x18ca7e0): adding session usage nta.c:4417 nta_leg_tcreate() nta_leg_tcreate(0x7f307c0bfdd0) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f307c0b7770) called soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7f307c0b7770, 0) called soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7f307c0b7770, soa_generate_offer): called soa_static.c:1189 offer_answer_step() soa_static(0x7f307c0b7770, soa_generate_offer): generating local description soa_static.c:1217 offer_answer_step() soa_static(0x7f307c0b7770, soa_generate_offer): upgrade with local description soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7f308b4e49f0, (nil), ""): called soa_static.c:1446 offer_answer_step() soa_static(0x7f307c0b7770, soa_generate_offer): storing local description soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f307c0b7770, [(nil)], [0x7f308b4e6b28], [0x7f308b4e6b24]) called nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip tport.c:4604 tport_by_name() tport(0x7f307c048c20): found 0x7f307c064180 by name tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3273 tport_tsend() tport_tsend(0x7f307c064180) tpn = tls/[2a01:7b30:a00f:62ba::3]:9999 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f307c061870 0x7f307c0f0750 911 (911) tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f307c061870 0x7f307c05d000 332 (332) tport.c:3610 tport_vsend() tport_vsend(0x7f307c064180): 1243 bytes of 1243 to tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3508 tport_send_msg() tport_vsend returned 1243 tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer nta.c:8310 outgoing_send() nta: sent INVITE (16621716) to tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:4176 tport_pend() tport_pend(0x7f307c064180): pending 0x7f307c0ec980 for tls/[2a01:7b30:a00f:62ba::3]:9999 (already 1) nua_session.c:4143 signal_call_state_change() nua(0x18ca7e0): call state changed: init -> calling, sent offer soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f307c0b7770, [0x7f308b4e6b08], [0x7f308b4e6b10], [(nil)]) called nua_stack.c:269 nua_stack_event() nua(0x18ca7e0): event i_state INVITE sent nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0x18ca7e0): sent signal r_invite 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:40 sofia/external-ipv6/9197 Standard INIT 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:48 (sofia/external-ipv6/9197) State Change CS_INIT -> CS_ROUTING 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:628 (sofia/external-ipv6/9197) State INIT going to sleep 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:585 (sofia/external-ipv6/9197) Running State Change CS_ROUTING (Cur 2 Tot 10) nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:56.887055 [DEBUG] sofia.c:7301 Channel sofia/external-ipv6/9197 entering state [calling][0] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:644 (sofia/external-ipv6/9197) State ROUTING 2020-02-22 08:56:56.887055 [DEBUG] mod_sofia.c:154 sofia/external-ipv6/9197 SOFIA ROUTING 2020-02-22 08:56:56.887055 [DEBUG] switch_ivr_originate.c:67 (sofia/external-ipv6/9197) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:644 (sofia/external-ipv6/9197) State ROUTING going to sleep 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:585 (sofia/external-ipv6/9197) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 10) 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:663 (sofia/external-ipv6/9197) State CONSUME_MEDIA 2020-02-22 08:56:56.887055 [DEBUG] switch_core_state_machine.c:663 (sofia/external-ipv6/9197) State CONSUME_MEDIA going to sleep tport.c:2782 tport_wakeup() tport_wakeup(0x7f307c064180): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f307c064180) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f307c064180): tls_read() returned 401 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f307c064180) msg 0x7f307c0acc80 from (tls/[2a01:7b30:a00f:62ba::3]:9999) has 401 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f307c064180): msg 0x7f307c0acc80 (401 bytes) from tls/[2a01:7b30:a00f:62ba::3]:9999/sips next=(nil) nta.c:3299 agent_recv_response() nta: received 100 Trying for INVITE (16621716) nta.c:3366 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9570 outgoing_estimate_delay() nta_outgoing: RTT is 23.315 ms tport.c:4238 tport_release() tport_release(0x7f307c064180): 0x7f307c0ec980 by 0x7f307c0b1410 with 0x7f307c0acc80 (preliminary) tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer tport.c:2782 tport_wakeup() tport_wakeup(0x7f307c064180): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f307c064180) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f307c064180): tls_read() returned 955 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f307c064180) msg 0x7f307c0acc80 from (tls/[2a01:7b30:a00f:62ba::3]:9999) has 955 bytes, veclen = 1 tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer tport.c:2782 tport_wakeup() tport_wakeup(0x7f307c064180): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f307c064180) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f307c064180): tls_read() returned 340 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f307c064180) msg 0x7f307c0acc80 from (tls/[2a01:7b30:a00f:62ba::3]:9999) has 340 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f307c064180): msg 0x7f307c0acc80 (1295 bytes) from tls/[2a01:7b30:a00f:62ba::3]:9999/sips next=(nil) nta.c:3299 agent_recv_response() nta: received 200 OK for INVITE (16621716) nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction tport.c:4238 tport_release() tport_release(0x7f307c064180): 0x7f307c0ec980 by 0x7f307c0b1410 with 0x7f307c0acc80 soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0x7f307c0b7770, (nil), 0x7f307c0ed3b0, 340) called soa.c:1595 soa_process_answer() soa_process_answer(static::0x7f307c0b7770) called soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7f307c0b7770, soa_process_answer): called soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7f309804da10, 0x7f308c016280, ""): called soa_static.c:1304 offer_answer_step() soa_static(0x7f307c0b7770, soa_process_answer): upgrade codecs with remote description soa.c:1730 soa_activate() soa_activate(static::0x7f307c0b7770, (nil)) called nua_session.c:992 nua_session_client_response() nua(0x18ca7e0): INVITE: processed SDP answer in 200 OK (200) nua_stack.c:271 nua_stack_event() nua(0x18ca7e0): event r_invite 200 OK nua_session.c:4143 signal_call_state_change() nua(0x18ca7e0): call state changed: calling -> completing, received answer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0x7f307c0b7770, [0x7f308b4e6558], [0x7f308b4e6560], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7f307c0b7770, ...) called nua_stack.c:271 nua_stack_event() nua(0x18ca7e0): event i_state 200 OK tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:57.927086 [INFO] sofia.c:1369 sofia/external-ipv6/9197 Update Callee ID to "9197" nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:57.927086 [DEBUG] sofia.c:7301 Channel sofia/external-ipv6/9197 entering state [completing][200] 2020-02-22 08:56:57.927086 [DEBUG] sofia.c:7311 Remote SDP: v=0 o=FreeSWITCH 1582325031 1582325032 IN IP6 2a01:7b30:a00f:62ba::3 s=FreeSWITCH c=IN IP6 2a01:7b30:a00f:62ba::3 t=0 0 m=audio 25986 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AEAD_AES_256_GCM_8 inline:YtCNFeeP7Nfe5QM64JWmcNBBdpwXAYEG3ZJ2e0aKTFJBhhiR4CCieyro68E= nua.c:639 nua_ack() nua: nua_ack: entering nua_stack.c:569 nua_stack_signal() nua(0x18ca7e0): recv signal r_ack nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f307c0b7770, ...) called soa.c:1730 soa_activate() soa_activate(static::0x7f307c0b7770, (nil)) called nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip tport.c:4604 tport_by_name() tport(0x7f307c048c20): found 0x7f307c064180 by name tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3273 tport_tsend() tport_tsend(0x7f307c064180) tpn = tls/[2a01:7b30:a00f:62ba::3]:9999 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f307c061870 0x7f307c0b4a50 461 (461) tport.c:3610 tport_vsend() tport_vsend(0x7f307c064180): 461 bytes of 461 to tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3508 tport_send_msg() tport_vsend returned 461 tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer nta.c:8310 outgoing_send() nta: sent ACK (16621716) to tls/[2a01:7b30:a00f:62ba::3]:9999 nua_session.c:4143 signal_call_state_change() nua(0x18ca7e0): call state changed: completing -> ready nua_stack.c:271 nua_stack_event() nua(0x18ca7e0): event i_state 200 ACK sent nua_stack.c:271 nua_stack_event() nua(0x18ca7e0): event i_active 200 Call active nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0x18ca7e0): sent signal r_ack nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:57.927086 [DEBUG] sofia.c:7301 Channel sofia/external-ipv6/9197 entering state [ready][200] 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:1813 looking for crypto suite [AEAD_AES_256_GCM_8]alias=[] in [1 AEAD_AES_256_GCM_8 inline:YtCNFeeP7Nfe5QM64JWmcNBBdpwXAYEG3ZJ2e0aKTFJBhhiR4CCieyro68E=] 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:1822 Found suite AEAD_AES_256_GCM_8 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:1892 Set Remote Key [1 AEAD_AES_256_GCM_8 inline:YtCNFeeP7Nfe5QM64JWmcNBBdpwXAYEG3ZJ2e0aKTFJBhhiR4CCieyro68E=] 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101 at 8000 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external-ipv6/9197 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-22 08:56:57.927086 [DEBUG] switch_core_codec.c:111 sofia/external-ipv6/9197 Original read codec set to PCMA:8 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101 at 8000 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:5909 sofia/external-ipv6/9197 Set 2833 dtmf send payload to 101 recv payload to 101 2020-02-22 08:56:57.927086 [DEBUG] switch_core_media.c:8659 AUDIO RTP [sofia/external-ipv6/9197] 2a2a:810b:a2ba::65 port 17212 -> 2a01:7b30:a00f:62ba::3 port 25986 codec: 8 ms: 20 2020-02-22 08:56:57.927086 [DEBUG] switch_rtp.c:4430 Not using a timer 2020-02-22 08:56:57.947037 [DEBUG] switch_core_media.c:8973 sofia/external-ipv6/9197 Set 2833 dtmf send payload to 101 2020-02-22 08:56:57.947037 [DEBUG] switch_core_media.c:8980 sofia/external-ipv6/9197 Set 2833 dtmf receive payload to 101 2020-02-22 08:56:57.947037 [DEBUG] switch_core_media.c:9003 sofia/external-ipv6/9197 Set rtp dtmf delay to 40 2020-02-22 08:56:57.947037 [INFO] switch_rtp.c:4212 Activating audio Secure RTP SEND 2020-02-22 08:56:57.947037 [DEBUG] switch_core_sqldb.c:2827 Secure Type: srtp:sdes:AEAD_AES_256_GCM_8 2020-02-22 08:56:57.947037 [INFO] switch_rtp.c:4190 Activating audio Secure RTP RECV 2020-02-22 08:56:57.947037 [DEBUG] switch_core_sqldb.c:2827 Secure Type: srtp:sdes:AEAD_AES_256_GCM_8 2020-02-22 08:56:57.947037 [NOTICE] sofia.c:8479 Channel [sofia/external-ipv6/9197] has been answered 2020-02-22 08:56:57.947037 [DEBUG] switch_channel.c:3865 (sofia/external-ipv6/9197) Callstate Change DOWN -> ACTIVE nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:1813 looking for crypto suite [AES_CM_128_HMAC_SHA1_80]alias=[] in [1 AES_CM_128_HMAC_SHA1_80 inline:djoxnaC5dQ7mre6tjhfK13Ca1sBgQKEIBjQab653|2^32] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:1822 Found suite AES_CM_128_HMAC_SHA1_80 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:1892 Set Remote Key [1 AES_CM_128_HMAC_SHA1_80 inline:djoxnaC5dQ7mre6tjhfK13Ca1sBgQKEIBjQab653|2^32] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:1215 Set Local audio crypto Key [1 AES_CM_128_HMAC_SHA1_80 inline:1FV79mvFW9EJ0Kzoz54tK71JzkmM8BYeOYcUd+uP] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G723:4:8000:20:6300:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [iLBC:97:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729E:102:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G726-32:2:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [AAL2-G726-40:103:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101 at 8000 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:3837 Set Codec sofia/external/1000 at 6.27.9.120:5061 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2020-02-22 08:56:57.967059 [DEBUG] switch_core_codec.c:111 sofia/external/1000 at 6.27.9.120:5061 Original read codec set to PCMA:8 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101 at 8000 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:5909 sofia/external/1000 at 6.27.9.120:5061 Set 2833 dtmf send payload to 101 recv payload to 101 2020-02-22 08:56:57.967059 [DEBUG] switch_core_media.c:8659 AUDIO RTP [sofia/external/1000 at 6.27.9.120:5061] 6.27.9.120 port 27568 -> 192.168.2.5 port 5012 codec: 8 ms: 20 2020-02-22 08:56:57.967059 [DEBUG] switch_rtp.c:4408 Starting timer [timerfd] 160 bytes per 20ms nta.c:1289 agent_timer() nta: timer not set 2020-02-22 08:56:59.027038 [DEBUG] switch_core_media.c:8973 sofia/external/1000 at 6.27.9.120:5061 Set 2833 dtmf send payload to 101 2020-02-22 08:56:59.027038 [DEBUG] switch_core_media.c:8980 sofia/external/1000 at 6.27.9.120:5061 Set 2833 dtmf receive payload to 101 2020-02-22 08:56:59.027038 [DEBUG] switch_core_media.c:9003 sofia/external/1000 at 6.27.9.120:5061 Set rtp dtmf delay to 40 2020-02-22 08:56:59.027038 [DEBUG] switch_core_media.c:1501 LIFETIME found in |2^32, base 2 exp 32 2020-02-22 08:56:59.027038 [NOTICE] switch_core_media.c:1524 Skipping MKI due to empty index 2020-02-22 08:56:59.027038 [INFO] switch_rtp.c:4212 Activating audio Secure RTP SEND 2020-02-22 08:56:59.027038 [DEBUG] switch_core_sqldb.c:2827 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2020-02-22 08:56:59.027038 [INFO] switch_rtp.c:4190 Activating audio Secure RTP RECV 2020-02-22 08:56:59.027038 [DEBUG] switch_core_sqldb.c:2827 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 2020-02-22 08:56:59.027038 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/1000 at 6.27.9.120:5061! 2020-02-22 08:56:59.027038 [DEBUG] switch_channel.c:3565 (sofia/external/1000 at 6.27.9.120:5061) Callstate Change RINGING -> EARLY 2020-02-22 08:56:59.027038 [DEBUG] switch_core_media.c:8641 Audio params are unchanged for sofia/external/1000 at 6.27.9.120:5061. 2020-02-22 08:56:59.027038 [DEBUG] mod_sofia.c:898 Local SDP sofia/external/1000 at 6.27.9.120:5061: v=0 o=FreeSWITCH 1582323451 1582323452 IN IP4 6.27.9.120 s=FreeSWITCH c=IN IP4 6.27.9.120 t=0 0 m=audio 27568 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1FV79mvFW9EJ0Kzoz54tK71JzkmM8BYeOYcUd+uP nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:573 nua_stack_signal() nua(0x7f3080063290): recv signal r_respond 200 OK nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800d65b0, ...) called soa.c:1052 soa_set_user_sdp() soa_set_user_sdp(static::0x7f30800d65b0, (nil), 0x7f309c050042, -1) called soa.c:890 soa_set_capability_sdp() soa_set_capability_sdp(static::0x7f30800d65b0, (nil), 0x7f309c050042, -1) called nua_session.c:2324 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1515 soa_generate_answer() soa_generate_answer(static::0x7f30800d65b0) called soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7f30800d65b0, soa_generate_answer): called soa_static.c:1189 offer_answer_step() soa_static(0x7f30800d65b0, soa_generate_answer): generating local description soa_static.c:1230 offer_answer_step() soa_static(0x7f30800d65b0, soa_generate_answer): upgrade with remote description soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7f30a14f7a20, 0x7f30800e2250, ""): called soa_static.c:1446 offer_answer_step() soa_static(0x7f30800d65b0, soa_generate_answer): storing local description soa.c:1730 soa_activate() soa_activate(static::0x7f30800d65b0, (nil)) called soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f30800d65b0, [(nil)], [0x7f30a14f9b58], [0x7f30a14f9b54]) called tport.c:3273 tport_tsend() tport_tsend(0x7f30800a2f80) tpn = TLS/33.2.151.64:2210 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f30800a3420 0x7f3080068f70 758 (758) tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f30800a3420 0x7f30800a0e50 305 (305) tport.c:3610 tport_vsend() tport_vsend(0x7f30800a2f80): 1063 bytes of 1063 to tls/33.2.151.64:2210 tport.c:3508 tport_send_msg() tport_vsend returned 1063 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:6797 incoming_reply() nta: sent 200 OK for INVITE (50) nta.c:1350 set_timeout() nta: timer set to 500 ms nua_session.c:4143 signal_call_state_change() nua(0x7f3080063290): call state changed: received -> completed, sent answer soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f30800d65b0, [0x7f30a14f9c08], [0x7f30a14f9c10], [(nil)]) called soa.c:616 soa_get_params() soa_get_params(static::0x7f30800d65b0, ...) called nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_state 200 OK nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0x7f3080063290): sent signal r_respond nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:59.027038 [DEBUG] sofia.c:7301 Channel sofia/external/1000 at 6.27.9.120:5061 entering state [completed][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:59.027038 [NOTICE] switch_ivr_originate.c:3751 Channel [sofia/external/1000 at 6.27.9.120:5061] has been answered 2020-02-22 08:56:59.027038 [DEBUG] switch_channel.c:3865 (sofia/external/1000 at 6.27.9.120:5061) Callstate Change EARLY -> ACTIVE 2020-02-22 08:56:59.027038 [DEBUG] switch_ivr_originate.c:3809 Originate Resulted in Success: [sofia/external-ipv6/9197] 2020-02-22 08:56:59.027038 [DEBUG] switch_ivr_bridge.c:1796 (sofia/external-ipv6/9197) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2020-02-22 08:56:59.027038 [DEBUG] switch_core_state_machine.c:585 (sofia/external-ipv6/9197) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 10) 2020-02-22 08:56:59.027038 [DEBUG] switch_core_state_machine.c:654 (sofia/external-ipv6/9197) State EXCHANGE_MEDIA 2020-02-22 08:56:59.027038 [DEBUG] mod_sofia.c:656 SOFIA EXCHANGE_MEDIA 2020-02-22 08:56:59.027038 [DEBUG] switch_rtp.c:7720 Correct audio ip/port confirmed. tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 568 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800f0fc0 from (tls/33.2.151.64:2210) has 568 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f30800a2f80): msg 0x7f30800f0fc0 (568 bytes) from tls/33.2.151.64:2210/sips next=(nil) nta.c:2880 agent_recv_request() nta: received ACK sip:009197 at 6.27.9.120:6666;transport=tls SIP/2.0 (CSeq 50) nta.c:3174 agent_check_request_via() nta: Via check: received=33.2.151.64 nta.c:3019 agent_recv_request() nta: ACK (50) is going to INVITE (50) nua_session.c:2573 process_ack_or_cancel() nua: process_ack_or_cancel: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0x7f30800d65b0) called nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_ack 200 OK nua_session.c:4143 signal_call_state_change() nua(0x7f3080063290): call state changed: completed -> ready nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_state 200 OK nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_active 200 Call active nta.c:5744 incoming_free() nta: incoming_free(0x7f30800cc750) tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:59.127093 [DEBUG] sofia.c:7301 Channel sofia/external/1000 at 6.27.9.120:5061 entering state [ready][200] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:798 nua_info() nua: nua_info: entering nua_stack.c:569 nua_stack_signal() nua(0x18ca7e0): recv signal r_info nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f307c0b7770, ...) called nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip tport.c:4604 tport_by_name() tport(0x7f307c048c20): found 0x7f307c064180 by name tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3273 tport_tsend() tport_tsend(0x7f307c064180) tpn = tls/[2a01:7b30:a00f:62ba::3]:9999 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f307c061870 0x7f307c0f9130 764 (764) tport.c:3610 tport_vsend() tport_vsend(0x7f307c064180): 764 bytes of 764 to tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:3508 tport_send_msg() tport_vsend returned 764 tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer nta.c:8310 outgoing_send() nta: sent INFO (16621717) to tls/[2a01:7b30:a00f:62ba::3]:9999 tport.c:4176 tport_pend() tport_pend(0x7f307c064180): pending 0x7f307c0f0ed0 for tls/[2a01:7b30:a00f:62ba::3]:9999 (already 1) nua_stack.c:529 nua_signal() nua(0x18ca7e0): sent signal r_info nua.c:810 nua_update() nua: nua_update: entering nua_stack.c:569 nua_stack_signal() nua(0x7f3080063290): recv signal r_update nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800d65b0, ...) called soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f30800d65b0) called soa.c:1426 soa_generate_offer() soa_generate_offer(static::0x7f30800d65b0, 0) called soa_static.c:1148 offer_answer_step() soa_static_offer_answer_action(0x7f30800d65b0, soa_generate_offer): called soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7f30800d99f0, (nil), ""): called soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f30800d65b0, [(nil)], [0x7f30a14f9b78], [0x7f30a14f9b74]) called nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip tport.c:3273 tport_tsend() tport_tsend(0x7f3080049a00) tpn = tls/192.168.2.5:5099 tport.c:4062 tport_resolve() tport_resolve addrinfo = 192.168.2.5:5099 tport.c:4696 tport_by_addrinfo() tport_by_addrinfo(0x7f3080049a00): not found by name tls/192.168.2.5:5099 tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f3080049a00): new secondary tport 0x7f30800e5000 tport_type_tcp.c:203 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7f30800e5000): Setting TCP_KEEPIDLE to 30 tport_type_tcp.c:209 tport_tcp_init_secondary() tport_tcp_init_secondary(0x7f30800e5000): Setting TCP_KEEPINTVL to 30 tport_type_tls.c:686 tport_tls_connect() tport_tls_connect(0x7f30800e5000): connecting to tls/192.168.2.5:5099/sips tport.c:2302 tport_set_secondary_timer() tport(0x7f30800e5000): reset timer tport.c:3798 tport_queue() tport_queue(0x7f30800e5000): queueing 0x7f30800e2e50 for tls/192.168.2.5:5099 nta.c:8310 outgoing_send() nta: sent UPDATE (16621717) to tls/192.168.2.5:5099 tport.c:4176 tport_pend() tport_pend(0x7f30800e5000): pending 0x7f30800e2e50 for tls/192.168.2.5:5099 (already 0) nua_session.c:4149 signal_call_state_change() nua(0x7f3080063290): ready call updated: calling sent offer soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7f30800d65b0, [0x7f30a14f9b58], [0x7f30a14f9b60], [(nil)]) called nua_stack.c:269 nua_stack_event() nua(0x7f3080063290): event i_state UPDATE sent nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0x7f3080063290): sent signal r_update nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:59.147097 [DEBUG] sofia.c:7301 Channel sofia/external/1000 at 6.27.9.120:5061 entering state [calling][0] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering tport.c:2782 tport_wakeup() tport_wakeup(0x7f307c064180): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f307c064180) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f307c064180): tls_read() returned 537 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f307c064180) msg 0x7f307c0f7420 from (tls/[2a01:7b30:a00f:62ba::3]:9999) has 537 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f307c064180): msg 0x7f307c0f7420 (537 bytes) from tls/[2a01:7b30:a00f:62ba::3]:9999/sips next=(nil) nta.c:3299 agent_recv_response() nta: received 200 OK for INFO (16621717) nta.c:3366 agent_recv_response() nta: 200 OK is going to a transaction nta.c:9570 outgoing_estimate_delay() nta_outgoing: RTT is 20.353 ms tport.c:4238 tport_release() tport_release(0x7f307c064180): 0x7f307c0f0ed0 by 0x7f307c0ac960 with 0x7f307c0f7420 nua_stack.c:271 nua_stack_event() nua(0x18ca7e0): event r_info 200 OK nta.c:8728 outgoing_free() nta: outgoing_free(0x7f307c0ac960) tport.c:2302 tport_set_secondary_timer() tport(0x7f307c064180): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:56:59.327055 [INFO] switch_rtp.c:7680 Auto Changing audio port from 192.168.2.5:5012 to 33.2.151.64:5012 nta.c:1296 agent_timer() nta: timer set next to 31631 ms tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 788 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800688b0 from (tls/33.2.151.64:2210) has 788 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f30800a2f80): msg 0x7f30800688b0 (788 bytes) from tls/33.2.151.64:2210/sips next=(nil) nta.c:2880 agent_recv_request() nta: received REGISTER sip:6.27.9.120:5061 SIP/2.0 (CSeq 2247) nta.c:3174 agent_check_request_via() nta: Via check: received=33.2.151.64 nta.c:3085 agent_recv_request() nta: REGISTER (2247) going to a default leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0x7f3080044490, 0x7f3080043c90, 0x7f30800aa610) called soa.c:403 soa_set_params() soa_set_params(static::0x7f30800eab50, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f30800aa610): adding registrar usage tport.c:4176 tport_pend() tport_pend(0x7f30800a2f80): pending (nil) for tls/33.2.151.64:2210 (already 0) nua_stack.c:271 nua_stack_event() nua(0x7f30800aa610): event i_register 100 Trying tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:573 nua_stack_signal() nua(0x7f30800aa610): recv signal r_respond 200 OK nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800eab50, ...) called tport.c:3273 tport_tsend() tport_tsend(0x7f30800a2f80) tpn = TLS/33.2.151.64:2210 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f30800a3420 0x7f30800e66e0 621 (621) tport.c:3610 tport_vsend() tport_vsend(0x7f30800a2f80): 621 bytes of 621 to tls/33.2.151.64:2210 tport.c:3508 tport_send_msg() tport_vsend returned 621 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:6797 incoming_reply() nta: sent 200 OK for REGISTER (2247) nta.c:5744 incoming_free() nta: incoming_free(0x7f30800cc750) nua_stack.c:529 nua_signal() nua(0x7f30800aa610): sent signal r_respond nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:569 nua_stack_signal() nua(0x7f30800aa610): recv signal r_destroy nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f30800aa610): removing registrar usage tport.c:4238 tport_release() tport_release(0x7f30800a2f80): (nil) by 0x7f30800aa610 with (nil) tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil)) soa.c:356 soa_destroy() soa_destroy(static::0x7f30800eab50) called nua_stack.c:529 nua_signal() nua(0x7f30800aa610): sent signal r_destroy nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nta.c:1296 agent_timer() nta: timer set next to 1052 ms nta.c:9107 outgoing_timer_dk() nta: timer D fired, terminate INVITE (16621703) nta.c:8805 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f308b4e6c60) nta.c:8935 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/2 tout, 1/2 term, 1/4 free nta.c:1296 agent_timer() nta: timer set next to 3 ms nta.c:8988 outgoing_timer_bf() nta: timer F fired, terminating ACK (16621703) nta.c:8805 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f308b4e6c60) nta.c:8935 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/2 tout, 0/1 term, 1/3 free nta.c:1296 agent_timer() nta: timer set next to 26697 ms tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 4 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800688b0 from (tls/33.2.151.64:2210) has 4 bytes, veclen = 1 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 788 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800688b0 from (tls/33.2.151.64:2210) has 788 bytes, veclen = 1 tport.c:3039 tport_deliver() tport_deliver(0x7f30800a2f80): msg 0x7f30800688b0 (792 bytes) from tls/33.2.151.64:2210/sips next=(nil) nta.c:2880 agent_recv_request() nta: received REGISTER sip:6.27.9.120:5061 SIP/2.0 (CSeq 2248) nta.c:3174 agent_check_request_via() nta: Via check: received=33.2.151.64 nta.c:3085 agent_recv_request() nta: REGISTER (2248) going to a default leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0x7f3080044490, 0x7f3080043c90, 0x7f30800ead90) called soa.c:403 soa_set_params() soa_set_params(static::0x7f30800eb600, ...) called nua_dialog.c:338 nua_dialog_usage_add() nua(0x7f30800ead90): adding registrar usage tport.c:4176 tport_pend() tport_pend(0x7f30800a2f80): pending (nil) for tls/33.2.151.64:2210 (already 0) nua_stack.c:271 nua_stack_event() nua(0x7f30800ead90): event i_register 100 Trying tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_stack.c:573 nua_stack_signal() nua(0x7f30800ead90): recv signal r_respond 200 OK nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0x7f30800eb600, ...) called tport.c:3273 tport_tsend() tport_tsend(0x7f30800a2f80) tpn = TLS/33.2.151.64:2210 tport_type_tls.c:537 tport_tls_send() tport_tls_writevec: vec 0x7f30800a3420 0x7f3080069a90 621 (621) tport.c:3610 tport_vsend() tport_vsend(0x7f30800a2f80): 621 bytes of 621 to tls/33.2.151.64:2210 tport.c:3508 tport_send_msg() tport_vsend returned 621 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:6797 incoming_reply() nta: sent 200 OK for REGISTER (2248) nta.c:5744 incoming_free() nta: incoming_free(0x7f30800b2d00) nua_stack.c:529 nua_signal() nua(0x7f30800ead90): sent signal r_respond nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:569 nua_stack_signal() nua(0x7f30800ead90): recv signal r_destroy nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f30800ead90): removing registrar usage tport.c:4238 tport_release() tport_release(0x7f30800a2f80): (nil) by 0x7f30800ead90 with (nil) tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:4470 nta_leg_destroy() nta_leg_destroy((nil)) soa.c:356 soa_destroy() soa_destroy(static::0x7f30800eb600) called nua_stack.c:529 nua_signal() nua(0x7f30800ead90): sent signal r_destroy nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering tport.c:2782 tport_wakeup() tport_wakeup(0x7f30800a2f80): events IN tport.c:2880 tport_recv_event() tport_recv_event(0x7f30800a2f80) tport_type_tls.c:434 tport_tls_recv() tport_tls_recv(0x7f30800a2f80): tls_read() returned 4 tport.c:3221 tport_recv_iovec() tport_recv_iovec(0x7f30800a2f80) msg 0x7f30800eb490 from (tls/33.2.151.64:2210) has 4 bytes, veclen = 1 tport.c:2302 tport_set_secondary_timer() tport(0x7f30800a2f80): reset timer nta.c:9107 outgoing_timer_dk() nta: timer D fired, terminate INVITE (16621716) nta.c:8805 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f308b4e6c60) nta.c:8935 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 0/1 tout, 1/1 term, 1/2 free nta.c:1296 agent_timer() nta: timer set next to 2 ms nta.c:8988 outgoing_timer_bf() nta: timer F fired, terminating ACK (16621716) nta.c:8805 outgoing_reclaim_queued() outgoing_reclaim_all((nil), (nil), 0x7f308b4e6c60) nta.c:8935 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set nta.c:8988 outgoing_timer_bf() nta: timer F fired, timeout UPDATE (16621717) tport.c:4238 tport_release() tport_release(0x7f30800e5000): 0x7f30800e2e50 by 0x7f30800f19f0 with (nil) nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event r_update 408 Request Timeout nta.c:8728 outgoing_free() nta: outgoing_free(0x7f30800f19f0) tport.c:2302 tport_set_secondary_timer() tport(0x7f30800e5000): reset timer nua_params.c:482 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:1784 soa_terminate() soa_terminate(static::0x7f30800d65b0) called soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0x7f30800d65b0) called nta.c:2665 nta_tpn_by_url() nta: selecting scheme sip tport.c:4604 tport_by_name() tport(0x7f3080049a00): found 0x7f30800e5000 by name tls/192.168.2.5:5099 tport.c:3273 tport_tsend() tport_tsend(0x7f30800e5000) tpn = tls/192.168.2.5:5099 tport.c:3798 tport_queue() tport_queue(0x7f30800e5000): queueing 0x7f30800e2250 for tls/192.168.2.5:5099 nta.c:8310 outgoing_send() nta: sent BYE (16621718) to tls/192.168.2.5:5099 tport.c:4176 tport_pend() tport_pend(0x7f30800e5000): pending 0x7f30800e2250 for tls/192.168.2.5:5099 (already 0) nua_session.c:4143 signal_call_state_change() nua(0x7f3080063290): call state changed: ready -> terminating nua_stack.c:271 nua_stack_event() nua(0x7f3080063290): event i_state 487 BYE sent nta.c:8935 _nta_outgoing_timer() nta_outgoing_timer: 0/0 resent, 1/1 tout, 0/0 term, 0/1 free nta.c:1296 agent_timer() nta: timer set next to 32001 ms nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2020-02-22 08:57:31.167054 [DEBUG] sofia.c:7301 Channel sofia/external/1000 at 6.27.9.120:5061 entering state [terminating][487] 2020-02-22 08:57:31.167054 [NOTICE] sofia.c:8534 Hangup sofia/external/1000 at 6.27.9.120:5061 [CS_EXECUTE] [ORIGINATOR_CANCEL] nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nua_stack.c:569 nua_stack_signal() nua(0x7f3080063290): recv signal r_destroy nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0x7f3080063290): removing session usage nua_session.c:4143 signal_call_state_change() nua(0x7f3080063290): call state changed: terminating -> terminated nua_stack.c:269 nua_stack_event() nua(0x7f3080063290): event i_state Terminated nua_stack.c:269 nua_stack_event() nua(0x7f3080063290): event i_terminated Terminated soa.c:356 soa_destroy() soa_destroy(static::0x7f30800d65b0) called nta.c:4470 nta_leg_destroy() nta_leg_destroy(0x7f30800f2c50) nua_stack.c:529 nua_signal() nua(0x7f3080063290): sent signal r_destroy nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering 2020-02-22 08:57:31.167054 [DEBUG] switch_ivr_bridge.c:829 sofia/external/1000 at 6.27.9.120:5061 ending bridge by request from read function 2020-02-22 08:57:31.167054 [DEBUG] switch_ivr_bridge.c:915 BRIDGE THREAD DONE [sofia/external/1000 at 6.27.9.120:5061] 2020-02-22 08:57:31.187101 [DEBUG] switch_ivr_bridge.c:823 sofia/external/1000 at 6.27.9.120:5061 ending bridge by request from write function 2020-02-22 08:57:31.187101 [DEBUG] switch_ivr_bridge.c:915 BRIDGE THREAD DONE [sofia/external-ipv6/9197] 2020-02-22 08:57:31.187101 [NOTICE] switch_ivr_bridge.c:1032 Hangup sofia/external-ipv6/9197 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] -- Regards, V.Bogatyryev -------------- next part -------------- An HTML attachment was scrubbed... URL: From iriik at yahoo.com Sat Feb 22 09:13:40 2020 From: iriik at yahoo.com (I K) Date: Sat, 22 Feb 2020 09:13:40 +0000 (UTC) Subject: RTP not transmitting over NAT References: <273111008.6613567.1582362820635.ref@mail.yahoo.com> Message-ID: <273111008.6613567.1582362820635@mail.yahoo.com> Thanks for answering Faisal. I configured stun on my linphone clients but it didn't solve the problem. I've also disabled SIP ALG  on my router but didn't make any difference. In this scenario which I'm using bypass-media feature of Freeswitch, everything works fine when clients are on the same network. But it does not work if one is another another remote network such as mobile data. In you are using linphone mobile clients make sure you have added "stun.linphone.org" in account specific stun settings,If you need both ZRTP & NAT handling keep media in proxy mode. On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users wrote: ---------- Forwarded message ---------- From: I K To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) Subject: RTP not transmitting over NAT Hi every one, I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?! Things I've acheived so far:1- bypass-media = false and proxy-media = false:In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well. 2- bypass-media = false and proxy-media = true:audio works but video shows blank3- bypass-media = true and proxy-media = false:If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s. I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help. ---------- Forwarded message ---------- From: I K via FreeSWITCH-users To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) Subject: [Freeswitch-users] RTP not transmitting over NAT _________________________________________________________________________ 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 iriik at yahoo.com Sat Feb 22 10:19:29 2020 From: iriik at yahoo.com (I K) Date: Sat, 22 Feb 2020 10:19:29 +0000 (UTC) Subject: RTP not transmitting over NAT References: <2028467456.6627334.1582366769164.ref@mail.yahoo.com> Message-ID: <2028467456.6627334.1582366769164@mail.yahoo.com> Thanks for helping David Villasmil. My Freeswitch is not behind NAT. It has a public IP. But my clients are behind NATs. Some of them are on the same network and others are using mobile data. My problem is simple:1- If I set internal profile on clients, those which are on the same network can have voice and video call with ZRTP using bypass-media. 2- If I set external profile on clients and set inbound-zrtp-passthru to "true", all of them can have (even those using mobile data) can have voice call with ZRTP. I'm looking for a solution to provide voice and VIDEO calls with ZRTP on all networks, regardless of NAT. If your fs is behind nat, you should set sip-ext-ip and rtp-ext-ip. This is done so that when fs sends an offer out it puts the sip-ext-ip and rtp ip in that offer. On the client side set the stun On Fri, 21 Feb 2020 at 04:38, Faisal Hanif wrote: In you are using linphone mobile clients make sure you have added "stun.linphone.org" in account specific stun settings,If you need both ZRTP & NAT handling keep media in proxy mode. On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users wrote: ---------- Forwarded message ---------- From: I K To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) Subject: RTP not transmitting over NAT Hi every one, I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?! Things I've acheived so far:1- bypass-media = false and proxy-media = false:In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well. 2- bypass-media = false and proxy-media = true:audio works but video shows blank3- bypass-media = true and proxy-media = false:If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s. I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help. ---------- Forwarded message ---------- From: I K via FreeSWITCH-users To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) Subject: [Freeswitch-users] RTP not transmitting over NAT _________________________________________________________________________ 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 Villasmilemail: david.villasmil.work at gmail.comphone: +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 iriik at yahoo.com Sat Feb 22 10:32:29 2020 From: iriik at yahoo.com (I K) Date: Sat, 22 Feb 2020 10:32:29 +0000 (UTC) Subject: RTP not transmitting over NAT References: <537875665.6649396.1582367549354.ref@mail.yahoo.com> Message-ID: <537875665.6649396.1582367549354@mail.yahoo.com> Dear Sergey Safarov, As I said earlier there are several scenario that I'm working on hoping one of them to work. When I use internal profile and I set bypass-media to "true", all clients residing on the same network can have voice and video call with or without ZRTP and adding or removing video to/from call works excellent.But, when using internal profile if I try to call one of clients on another network such as mobile data, the call takes place but no audio is gransmitted(no ZRTP).And if I use external profile and set bypass-media and inbound-zrtp-passthru to "true", all clients regardless of NAT, Wi-fi or mobile data can call each other with ZRTP but no video! when I press the video button on Linphone nothing happens! and of course I've added H26x to codecs.I've also reviewed Jira topics and commits you mentioned but they are not related to my problem. Thanks again for helping. Is without ZRTP video/addition removing works well? Case 1Without ZRTP similar case fixed about two years ago. Sorry I cannot find related ticked created by me (jira login safarov). Please retest master and try bisect with commits 1-3 years ago. Think you will find working version and where bug is introduced. Case 2 Please look https://freeswitch.org/jira/browse/FS-9203 https://freeswitch.org/jira/browse/FS-9206 Here you will find commit where it worked properly. Just need bisect with master and find commit introduced bug. Case 3Useless for you. This not designed for users behind NAT. So simple do not use. Sergey On Fri, Feb 21, 2020 at 7:22 AM Faisal Hanif wrote: In you are using linphone mobile clients make sure you have added "stun.linphone.org" in account specific stun settings,If you need both ZRTP & NAT handling keep media in proxy mode. On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users wrote: ---------- Forwarded message ---------- From: I K To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) Subject: RTP not transmitting over NAT Hi every one, I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?! Things I've acheived so far:1- bypass-media = false and proxy-media = false:In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well. 2- bypass-media = false and proxy-media = true:audio works but video shows blank3- bypass-media = true and proxy-media = false:If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s. I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help. ---------- Forwarded message ---------- From: I K via FreeSWITCH-users To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) Subject: [Freeswitch-users] RTP not transmitting over NAT _________________________________________________________________________ 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 If your fs is behind nat, you should set sip-ext-ip and rtp-ext-ip. This is done so that when fs sends an offer out it puts the sip-ext-ip and rtp ip in that offer. On the client side set the stun On Fri, 21 Feb 2020 at 04:38, Faisal Hanif wrote: In you are using linphone mobile clients make sure you have added "stun.linphone.org" in account specific stun settings,If you need both ZRTP & NAT handling keep media in proxy mode. On Fri, Feb 21, 2020, 6:29 AM I K via FreeSWITCH-users wrote: ---------- Forwarded message ---------- From: I K To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Sat, 15 Feb 2020 22:11:12 +0000 (UTC) Subject: RTP not transmitting over NAT Hi every one, I'm new to Freeswitch and I'm having some difficulty configuring it, despite reading the documents. I have a Freeswitch running on a server with public IP and several Linphone android clients running behind NAT.How should I configure it to transfer audio and video and use ZRTP if one client is on a LAN behind NAT and the other client is using mobile data?! Things I've acheived so far:1- bypass-media = false and proxy-media = false:In this mode audio works but when I turn on video no request is sent to the other phone. It other use just has to know when to turn on the camera to receive video. and of course ZRTP does not work well. 2- bypass-media = false and proxy-media = true:audio works but video shows blank3- bypass-media = true and proxy-media = false:If both clients are on the same LAN everything works fine, but if one client connects via mobile data, they can still call each other but no media is transferred and the call is dropped after 30s. I've been working on this scenario for over a week and I'm desperate to make it work. Thanks in advance for your help. ---------- Forwarded message ---------- From: I K via FreeSWITCH-users To: "freeswitch-users at lists.freeswitch.org" Cc:  Bcc:  Date: Thu, 20 Feb 2020 17:29:02 -0800 (PST) Subject: [Freeswitch-users] RTP not transmitting over NAT _________________________________________________________________________ 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 Villasmilemail: david.villasmil.work at gmail.comphone: +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 mahmood.alkhalil at outlook.com Sun Feb 23 14:07:31 2020 From: mahmood.alkhalil at outlook.com (Mahmood Alkhalil) Date: Sun, 23 Feb 2020 14:07:31 +0000 Subject: [Freeswitch-users] JsSip Client receiving "488 Not Acceptable Here" when trying to make a call In-Reply-To: References: Message-ID: Hi Piotr/David, Thanks for you help. turns out JsSip send wrong SDP "m=" field, I've started using SipJS library and it's working fine without any issues. i was wondering if anyone is using SipJS (https://sipjs.com/) in production environment and can provide a feedback on its stability and quality, I'm planning to start building a project using SipJS library, so would really appreciate any feedback On 2/18/20 5:51 AM, David Villasmil wrote: Get a trace and check your codecs. On Mon, 17 Feb 2020 at 22:57, Piotr Gregor > wrote: Hi Mahmood, Make sure DTLS/ICE works. Check with tcpdump/wireshark packets are flowing between ends of the call. Define ICE servers to be used by JsSip. var options = { 'eventHandlers' : your eventHandlers, 'mediaConstraints' : { 'audio': true, 'video': true }, 'pcConfig' : { 'iceServers': [ {'urls': 'stun:stun.stunprotocol.org:3478'}, {'urls': 'stun:stun.l.google.com:19302'}, ] }, }; var session = ua.call(uri, options); kind regards, Piotr [https://drive.google.com/a/dataandsignal.com/uc?id=1lUdzdLjEycxXcGhDz-bKqDX8-AIlZuCq&export=download] Piotr Gregor Software Engineer M: (+44) 07483 866 525 www: dataandsignal.com On Wed, Feb 12, 2020 at 4:07 PM Mahmood Alkhalil > wrote: Hi Everyone, I've been trying to use JsSIp over Websockets, I'm able to register and i can confirm the registration is a success, but whenevr i try to make a call Freeswitch responds with SIP/2.0 488 Not Acceptable Here. ​The JsSip API is showing SDP Failure, in Freeswitch logs it is telling that there is SDP/RTP matches but then it say no candidate found. I'v attached the call trace, appreciate any help. Thanks, Mahmood Alkhalil. _________________________________________________________________________ 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 remi.marand at odigo.com Mon Feb 24 15:09:54 2020 From: remi.marand at odigo.com (MARAND, Remi) Date: Mon, 24 Feb 2020 15:09:54 +0000 Subject: [Freeswitch-users] Google Chrome Version 81. Message-ID: We use Freeswitch in the case of a SIP to WebRTC application and it is a solid and reliable solution. When we test calls with the sipjs.js library (V0.15.10) and Freeswitch (FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) + Debian 9, with the beta version of Chrome, Version 81.0.4044.17 (Official Build) beta (64-bit), we observe a notification on the javascript console : "[Deprecation] Your partner is negotiating an obsolete (D)TLS version. Support for this will be removed in M81, around March 2020. Please check with your partner to have this fixed. » The Wireshark trace shows that the DTLS-SRTP exchange is done with DTLSv1.0. What would be the version update to do on the Freeswitch side (or Debian) to avoid this deprecation ? Is there a configuration to do in our current version? Best regards. Rémi Marand Product Owner | Marketing & Product Odigo France | Issy les Moulineaux Tel.: +33 1 46 84 12 77 / +33 6 87 72 53 25. www.odigo.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From enp at itx.ru Mon Feb 24 17:08:14 2020 From: enp at itx.ru (Eugene Prokopiev) Date: Mon, 24 Feb 2020 20:08:14 +0300 Subject: [Freeswitch-users] WebRTC SSL termination on Nginx Message-ID: Hi, I tried to terminate WebRTC SSL before FreeSWITCH on Nginx in this way: server { server_name mydomain.com; listen 443 ssl; ssl_certificate /etc/nginx/certs/fullchain; ssl_certificate_key /etc/nginx/certs/key; ssl_trusted_certificate /etc/nginx/certs/cert; location / { proxy_pass http://10.0.0.10:5070; proxy_http_version 1.1; proxy_set_header Host $http_host; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; } } FreeSWITCH configuration looks like: But I see this error: nta.c:2880 agent_recv_request() nta: received REGISTER sip:10.0.0.10 SIP/2.0 (CSeq 1) nta.c:3146 agent_check_request_via() nta: Via check: invalid transport "SIP/2.0/WSS" from 10.0.0.1:40568 nta.c:2990 agent_recv_request() nta: REGISTER (1) has invalid Via No wonder really, it is not encripted WS between Nginx and FreeSWITCH, WSS with encription is before Nginx only, but WSS defined in REGISTER Via header. But with wss-binding param instead of ws-binding I see another error: tport.c:2622 tport_accept() tport_accept(0x7f8290005180): incoming secondary on wss/10.0.0.10:5070/sips failed. reason = WS_INIT No wonder again, FreeSWITCH waiting for WSS, but receiving not encriped WS But I'm confused. Maybe it is possible to rewrite Via header? Or FreeSWITCH can't work behind WSS proxy with not encripted WS between proxy and FreeSWITCH? -- WBR, Eugene Prokopiev From aidar.kamalov at gmail.com Tue Feb 25 09:25:42 2020 From: aidar.kamalov at gmail.com (Aidar Kamalov) Date: Tue, 25 Feb 2020 12:25:42 +0300 Subject: [Freeswitch-users] low level as ARI Message-ID: Hello, is it possible in freeswitch to use channels and bridges on low level as in ARI in asterisk? When external user call to freeswitch I need to create bridge and get it's uuid, then I want to move new created channel to new created bridge play greeting sound and start music-on-hold in bridge. After that I want to create outgoing call to freeswitch user and when he answered move his channel to already created bridge. Then this user can add to this bridge another users. So, I already can call to freeswitch, park this call, play greeting sound to caller, make outbound call to freeswitch user and park it too. But how to create empty bridge and how to add to this bridge existing channels? -- Aydar A. Kamalov -------------- next part -------------- An HTML attachment was scrubbed... URL: From remi.marand at odigo.com Fri Feb 28 12:32:02 2020 From: remi.marand at odigo.com (MARAND, Remi) Date: Fri, 28 Feb 2020 12:32:02 +0000 Subject: [Freeswitch-users] Help for Google Chrome version 81 & FS. Message-ID: We use Freeswitch in the case of a SIP to WebRTC application and it is a solid and reliable solution. When we test calls with the sipjs.js library (V0.15.10) and Freeswitch (FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) + Debian 9, with the beta version of Chrome, Version 81.0.4044.17 (Official Build) beta (64-bit), we observe a notification on the javascript console : "[Deprecation] Your partner is negotiating an obsolete (D)TLS version. Support for this will be removed in M81, around March 2020. Please check with your partner to have this fixed. » The Wireshark trace shows that the DTLS-SRTP exchange is done with DTLSv1.0. What would be the version update to do on the Freeswitch side (or Debian) to avoid this deprecation ? Is there a configuration to do in our current version? Best regards. Rémi Marand Odigo France | Issy les Moulineaux Tel.: +33 1 46 84 12 77 / +33 6 87 72 53 25. www.odigo.com This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Fri Feb 28 15:31:16 2020 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 28 Feb 2020 16:31:16 +0100 Subject: [Freeswitch-users] Google Chrome Version 81. In-Reply-To: References: Message-ID: <6074581F-FE0F-4DE4-B613-85DEF6051C6D@vallimamod.org> Hi, Current stable version of freeswitch is 1.10 (https://github.com/signalwire/freeswitch/tree/v1.10 ) on Debian 10. You should check with this version. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 24 Feb 2020, at 16:09, MARAND, Remi > wrote: > > > > We use Freeswitch in the case of a SIP to WebRTC application and it is a solid and reliable solution. > > When we test calls with the sipjs.js library (V0.15.10) and Freeswitch (FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) + Debian 9, with the beta version of Chrome, Version 81.0.4044.17 (Official Build) beta (64-bit), we observe a notification on the javascript console : “[Deprecation] Your partner is negotiating an obsolete (D)TLS version. Support for this will be removed in M81, around March 2020. Please check with your partner to have this fixed. » > > The Wireshark trace shows that the DTLS-SRTP exchange is done with DTLSv1.0. > > What would be the version update to do on the Freeswitch side (or Debian) to avoid this deprecation ? > Is there a configuration to do in our current version? > > Best regards. > > Rémi Marand > Product Owner | Marketing & Product > > Odigo France | Issy les Moulineaux > Tel.: +33 1 46 84 12 77 / +33 6 87 72 53 25. > www.odigo.com > > This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message. _________________________________________________________________________ > > 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 Bien cordialement, -- Vallimamod Abdullah SIP Solutions Conseil & Développement de plateformes et services VOIP vma at sip.solutions +33 6 62 60 68 97 linkedin.com/in/vallimamod . -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Feb 28 16:27:21 2020 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 28 Feb 2020 13:27:21 -0300 Subject: [Freeswitch-users] low level as ARI In-Reply-To: References: Message-ID: Are you using LUA? ESL? Here is a LUA example you can modify to your needs: https://freeswitch.org/confluence/display/FREESWITCH/Lua+example+Bridging+two+calls+with+retry In ESL, just bridge the two UUIDs together. You might also want to look up INTERCEPT in Confluence. Guillermo On Fri, Feb 28, 2020 at 12:55 PM Aidar Kamalov wrote: > Hello, > > is it possible in freeswitch to use channels and bridges on low level as > in ARI in asterisk? > > When external user call to freeswitch I need to create bridge and get it's > uuid, then I want to move new created channel to new created bridge play > greeting sound and start music-on-hold in bridge. > > After that I want to create outgoing call to freeswitch user and when he > answered move his channel to already created bridge. Then this user can add > to this bridge another users. > > So, I already can call to freeswitch, park this call, play greeting sound > to caller, make outbound call to freeswitch user and park it too. But how > to create empty bridge and how to add to this bridge existing channels? > > -- > Aydar A. Kamalov > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Fri Feb 28 16:28:39 2020 From: brian at freeswitch.com (Brian West) Date: Fri, 28 Feb 2020 10:28:39 -0600 Subject: [Freeswitch-users] Freeswitch 32bits In-Reply-To: References: Message-ID: We no longer support 32bit via packages. Thanks, Brian On Fri, Feb 28, 2020 at 9:04 AM John Tuxies wrote: > Trying to install freeswitch in a debian 10 32bits (ALIX 2d13) machine > from source and i am getting the following error: > > > > > > > > > > > > > > > > > > > *root at alix2d13: ~ $ apt-get build-dep freeswitchReading package lists... > DoneReading package lists... DoneBuilding dependency treeReading state > information... DoneSome packages could not be installed. This may mean that > you haverequested an impossible situation or if you are using the > unstabledistribution that some required packages have not yet been > createdor been moved out of Incoming.The following information may help to > resolve the situation:The following packages have unmet > dependencies: builddeps:freeswitch : Depends: libks but it is not > installable Depends: signalwire-client-c but it is > not installable Depends: libbroadvoice-dev but it is > not installable Depends: libsilk-dev but it is not > installable Depends: libv8-6.1-dev but it is not > installableE: Unable to correct problems, you have held broken > packages.root at alix2d13: ~ $* > > No idea how to overcome this issue. > > > _________________________________________________________________________ > > 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... 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