From loidang at hoiio.com Fri Nov 1 02:30:34 2019 From: loidang at hoiio.com (Loi Dang) Date: Fri, 1 Nov 2019 09:30:34 +0700 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: Hi, i experienced this 2 years ago with the latest version that time 1.6.17, likely no will to change for the current version. For short, proxy media does NOT seem to work at all for `t38-passthru=true`. My solution was to switch to normal media mode for that option to take effect. You should give it a try. rgds, Loi Dang On Fri, Nov 1, 2019 at 1:13 AM Brian West wrote: > Don't use proxy media mode, PRETEND PROXY MEDIA DOESN'T EXIST! :) > > On Thu, Oct 31, 2019 at 12:35 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Can you paste the complete messages? >> >> On Thu, 31 Oct 2019 at 16:44, Jose Fco. Irles Durá >> wrote: >> >>> Hi, I have sended another email to the list with attachments. I think >>> that the list not accepts attachments, sorry. >>> >>> I have several FreeSWITCH's (latest version from deb packages: 1.10.1 >>> -release-12-f9990221e6) and I have problems with faxes in proxy media >>> (in bypass-media works). >>> In the sofia.conf.xml I have setted the param t38-passthru=true >>> >>> If the flow is: >>> 1. Initial INVITE >>> 2. Re-Invite from one of the legs with t38 sdp >>> Result: works >>> >>> But if the flow is: >>> 1. Initial INVITE >>> 2. Re-invite from one of the legs changing the media (at answer) >>> 3. Re-invite from one of the other legs with t38 sdp >>> Result: not works >>> >>> I tried with "soa_enable=false" without luck. >>> >>> FreeSWITCH seems that hangs the sip transaction. >>> >>> This capture is the first reinvite (in the flow, the "showing" line): >>> https://pastebin.com/raw/X4zrUr4k >>> >>> This capture is the reinvite with the t38 negotiation sended by the >>> receiver (Grandstream ATA): >>> https://pastebin.com/raw/KCXD51aH >>> >>> And this with the response (200ok) accepting the t38 negotiation): >>> https://pastebin.com/raw/ifqtPN8L >>> >>> How I can debug the problem? >>> >>> Best regards >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> -- >> 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 > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Fri Nov 1 19:59:09 2019 From: brian at freeswitch.com (Brian West) Date: Fri, 1 Nov 2019 14:59:09 -0500 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: You should update. /b On Thu, Oct 31, 2019 at 9:31 PM Loi Dang wrote: > Hi, i experienced this 2 years ago with the latest version that time > 1.6.17, likely no will to change for the current version. For short, proxy > media does NOT seem to work at all for `t38-passthru=true`. > My solution was to switch to normal media mode for that option to take > effect. > You should give it a try. > > rgds, > Loi Dang > > On Fri, Nov 1, 2019 at 1:13 AM Brian West wrote: > >> Don't use proxy media mode, PRETEND PROXY MEDIA DOESN'T EXIST! :) >> >> On Thu, Oct 31, 2019 at 12:35 PM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Can you paste the complete messages? >>> >>> On Thu, 31 Oct 2019 at 16:44, Jose Fco. Irles Durá >>> wrote: >>> >>>> Hi, I have sended another email to the list with attachments. I think >>>> that the list not accepts attachments, sorry. >>>> >>>> I have several FreeSWITCH's (latest version from deb packages: 1.10.1 >>>> -release-12-f9990221e6) and I have problems with faxes in proxy media >>>> (in bypass-media works). >>>> In the sofia.conf.xml I have setted the param t38-passthru=true >>>> >>>> If the flow is: >>>> 1. Initial INVITE >>>> 2. Re-Invite from one of the legs with t38 sdp >>>> Result: works >>>> >>>> But if the flow is: >>>> 1. Initial INVITE >>>> 2. Re-invite from one of the legs changing the media (at answer) >>>> 3. Re-invite from one of the other legs with t38 sdp >>>> Result: not works >>>> >>>> I tried with "soa_enable=false" without luck. >>>> >>>> FreeSWITCH seems that hangs the sip transaction. >>>> >>>> This capture is the first reinvite (in the flow, the "showing" line): >>>> https://pastebin.com/raw/X4zrUr4k >>>> >>>> This capture is the reinvite with the t38 negotiation sended by the >>>> receiver (Grandstream ATA): >>>> https://pastebin.com/raw/KCXD51aH >>>> >>>> And this with the response (200ok) accepting the t38 negotiation): >>>> https://pastebin.com/raw/ifqtPN8L >>>> >>>> How I can debug the problem? >>>> >>>> Best regards >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> 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 >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From josefu at gmail.com Sat Nov 2 19:43:46 2019 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Sat, 2 Nov 2019 20:43:46 +0100 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: I think that i tested in the default mode and I receive from FreeSWITCH a 488, but I'm not sure at this moment. I will retest the next monday. Best regards El vie., 1 nov. 2019 a las 3:31, Loi Dang () escribió: > Hi, i experienced this 2 years ago with the latest version that time > 1.6.17, likely no will to change for the current version. For short, proxy > media does NOT seem to work at all for `t38-passthru=true`. > My solution was to switch to normal media mode for that option to take > effect. > You should give it a try. > > rgds, > Loi Dang > > On Fri, Nov 1, 2019 at 1:13 AM Brian West wrote: > >> Don't use proxy media mode, PRETEND PROXY MEDIA DOESN'T EXIST! :) >> >> On Thu, Oct 31, 2019 at 12:35 PM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Can you paste the complete messages? >>> >>> On Thu, 31 Oct 2019 at 16:44, Jose Fco. Irles Durá >>> wrote: >>> >>>> Hi, I have sended another email to the list with attachments. I think >>>> that the list not accepts attachments, sorry. >>>> >>>> I have several FreeSWITCH's (latest version from deb packages: 1.10.1 >>>> -release-12-f9990221e6) and I have problems with faxes in proxy media >>>> (in bypass-media works). >>>> In the sofia.conf.xml I have setted the param t38-passthru=true >>>> >>>> If the flow is: >>>> 1. Initial INVITE >>>> 2. Re-Invite from one of the legs with t38 sdp >>>> Result: works >>>> >>>> But if the flow is: >>>> 1. Initial INVITE >>>> 2. Re-invite from one of the legs changing the media (at answer) >>>> 3. Re-invite from one of the other legs with t38 sdp >>>> Result: not works >>>> >>>> I tried with "soa_enable=false" without luck. >>>> >>>> FreeSWITCH seems that hangs the sip transaction. >>>> >>>> This capture is the first reinvite (in the flow, the "showing" line): >>>> https://pastebin.com/raw/X4zrUr4k >>>> >>>> This capture is the reinvite with the t38 negotiation sended by the >>>> receiver (Grandstream ATA): >>>> https://pastebin.com/raw/KCXD51aH >>>> >>>> And this with the response (200ok) accepting the t38 negotiation): >>>> https://pastebin.com/raw/ifqtPN8L >>>> >>>> How I can debug the problem? >>>> >>>> Best regards >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> -- >>> 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 >> >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Jose Fco. Irles Durá -------------- next part -------------- An HTML attachment was scrubbed... URL: From josefu at gmail.com Mon Nov 4 08:19:30 2019 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Mon, 4 Nov 2019 09:19:30 +0100 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: Hi, I have retested in default mode and FreeSWITCH refuse t38: https://pastebin.com/raw/5LVvFdht In the console I can see: switch_core_media.c:4028 sofia/myprofile/number at 33.33.33.33 T38 REFUSE on request In my dialplay I set: bypass_media=false proxy_media=false t38_passthru=true (same result setted or not. In sofia profile is setted the param) fax_enable_t38=true (same result setted or not) I add to the dialplan a ringback with ringback=%(1500,3000,425) This is my sofia configuration (params section): https://pastebin.com/usW3AKa0 What is wrong in my config? Best regards El sáb., 2 nov. 2019 a las 20:43, Jose Fco. Irles Durá () escribió: > I think that i tested in the default mode and I receive from FreeSWITCH a > 488, but I'm not sure at this moment. I will retest the next monday. > > Best regards > > El vie., 1 nov. 2019 a las 3:31, Loi Dang () escribió: > >> Hi, i experienced this 2 years ago with the latest version that time >> 1.6.17, likely no will to change for the current version. For short, proxy >> media does NOT seem to work at all for `t38-passthru=true`. >> My solution was to switch to normal media mode for that option to take >> effect. >> You should give it a try. >> >> rgds, >> Loi Dang >> >> On Fri, Nov 1, 2019 at 1:13 AM Brian West wrote: >> >>> Don't use proxy media mode, PRETEND PROXY MEDIA DOESN'T EXIST! :) >>> >>> On Thu, Oct 31, 2019 at 12:35 PM David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>>> Can you paste the complete messages? >>>> >>>> On Thu, 31 Oct 2019 at 16:44, Jose Fco. Irles Durá >>>> wrote: >>>> >>>>> Hi, I have sended another email to the list with attachments. I think >>>>> that the list not accepts attachments, sorry. >>>>> >>>>> I have several FreeSWITCH's (latest version from deb packages: 1.10.1 >>>>> -release-12-f9990221e6) and I have problems with faxes in proxy media >>>>> (in bypass-media works). >>>>> In the sofia.conf.xml I have setted the param t38-passthru=true >>>>> >>>>> If the flow is: >>>>> 1. Initial INVITE >>>>> 2. Re-Invite from one of the legs with t38 sdp >>>>> Result: works >>>>> >>>>> But if the flow is: >>>>> 1. Initial INVITE >>>>> 2. Re-invite from one of the legs changing the media (at answer) >>>>> 3. Re-invite from one of the other legs with t38 sdp >>>>> Result: not works >>>>> >>>>> I tried with "soa_enable=false" without luck. >>>>> >>>>> FreeSWITCH seems that hangs the sip transaction. >>>>> >>>>> This capture is the first reinvite (in the flow, the "showing" line): >>>>> https://pastebin.com/raw/X4zrUr4k >>>>> >>>>> This capture is the reinvite with the t38 negotiation sended by the >>>>> receiver (Grandstream ATA): >>>>> https://pastebin.com/raw/KCXD51aH >>>>> >>>>> And this with the response (200ok) accepting the t38 negotiation): >>>>> https://pastebin.com/raw/ifqtPN8L >>>>> >>>>> How I can debug the problem? >>>>> >>>>> Best regards >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> The FreeSWITCH project is sponsored by SignalWire >>>>> https://signalwire.com >>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>> services. >>>>> Build your next product on our scalable cloud platform. >>>>> >>>>> Join our online community to chat in real time >>>>> https://signalwire.community >>>>> >>>>> Professional FreeSWITCH Services >>>>> sales at freeswitch.com >>>>> https://freeswitch.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> https://freeswitch.com/oss >>>>> https://freeswitch.org/confluence >>>>> https://cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> https://freeswitch.com >>>> >>>> -- >>>> 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 >>> >>> >>> >>> -- >>> >>> Brian West | Co-founder and Developer >>> >>> Need Commercial support? email sales at freeswitch.com >>> >>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>> >>> >>> Email: brian at freeswitch.com >>> >>> Mobile: 918-424-9378 >>> >>> Website: https://www.FreeSWITCH.com >>> >>> [image: https://www.facebook.com/signalwireinc?src=email] >>> [image: >>> https://twitter.com/freeswitch] >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Jose Fco. Irles Durá > -- Jose Fco. Irles Durá -------------- next part -------------- An HTML attachment was scrubbed... URL: From loidang at hoiio.com Mon Nov 4 10:25:45 2019 From: loidang at hoiio.com (Loi Dang) Date: Mon, 4 Nov 2019 17:25:45 +0700 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: Hi, any luck `fax_enable_t38` is only set in your dialplan, but not exported? As t38 are rejected at the outgoing leg. What is your FS version, anw? rgds, Loi Dang On Mon, Nov 4, 2019 at 4:09 PM Jose Fco. Irles Durá wrote: > Hi, > > I have retested in default mode and FreeSWITCH refuse t38: > https://pastebin.com/raw/5LVvFdht > > In the console I can see: > switch_core_media.c:4028 sofia/myprofile/number at 33.33.33.33 T38 REFUSE on > request > > In my dialplay I set: > bypass_media=false > proxy_media=false > t38_passthru=true (same result setted or not. In sofia profile is setted > the param) > fax_enable_t38=true (same result setted or not) > > I add to the dialplan a ringback with > ringback=%(1500,3000,425) > > This is my sofia configuration (params section): > https://pastebin.com/usW3AKa0 > > What is wrong in my config? > > Best regards > > El sáb., 2 nov. 2019 a las 20:43, Jose Fco. Irles Durá () > escribió: > >> I think that i tested in the default mode and I receive from FreeSWITCH a >> 488, but I'm not sure at this moment. I will retest the next monday. >> >> Best regards >> >> El vie., 1 nov. 2019 a las 3:31, Loi Dang () escribió: >> >>> Hi, i experienced this 2 years ago with the latest version that time >>> 1.6.17, likely no will to change for the current version. For short, proxy >>> media does NOT seem to work at all for `t38-passthru=true`. >>> My solution was to switch to normal media mode for that option to take >>> effect. >>> You should give it a try. >>> >>> rgds, >>> Loi Dang >>> >>> On Fri, Nov 1, 2019 at 1:13 AM Brian West wrote: >>> >>>> Don't use proxy media mode, PRETEND PROXY MEDIA DOESN'T EXIST! :) >>>> >>>> On Thu, Oct 31, 2019 at 12:35 PM David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Can you paste the complete messages? >>>>> >>>>> On Thu, 31 Oct 2019 at 16:44, Jose Fco. Irles Durá >>>>> wrote: >>>>> >>>>>> Hi, I have sended another email to the list with attachments. I think >>>>>> that the list not accepts attachments, sorry. >>>>>> >>>>>> I have several FreeSWITCH's (latest version from deb packages: 1.10.1 >>>>>> -release-12-f9990221e6) and I have problems with faxes in proxy media >>>>>> (in bypass-media works). >>>>>> In the sofia.conf.xml I have setted the param t38-passthru=true >>>>>> >>>>>> If the flow is: >>>>>> 1. Initial INVITE >>>>>> 2. Re-Invite from one of the legs with t38 sdp >>>>>> Result: works >>>>>> >>>>>> But if the flow is: >>>>>> 1. Initial INVITE >>>>>> 2. Re-invite from one of the legs changing the media (at answer) >>>>>> 3. Re-invite from one of the other legs with t38 sdp >>>>>> Result: not works >>>>>> >>>>>> I tried with "soa_enable=false" without luck. >>>>>> >>>>>> FreeSWITCH seems that hangs the sip transaction. >>>>>> >>>>>> This capture is the first reinvite (in the flow, the "showing" line): >>>>>> https://pastebin.com/raw/X4zrUr4k >>>>>> >>>>>> This capture is the reinvite with the t38 negotiation sended by the >>>>>> receiver (Grandstream ATA): >>>>>> https://pastebin.com/raw/KCXD51aH >>>>>> >>>>>> And this with the response (200ok) accepting the t38 negotiation): >>>>>> https://pastebin.com/raw/ifqtPN8L >>>>>> >>>>>> How I can debug the problem? >>>>>> >>>>>> Best regards >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> The FreeSWITCH project is sponsored by SignalWire >>>>>> https://signalwire.com >>>>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>>>> services. >>>>>> Build your next product on our scalable cloud platform. >>>>>> >>>>>> Join our online community to chat in real time >>>>>> https://signalwire.community >>>>>> >>>>>> Professional FreeSWITCH Services >>>>>> sales at freeswitch.com >>>>>> https://freeswitch.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> https://freeswitch.com/oss >>>>>> https://freeswitch.org/confluence >>>>>> https://cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> https://freeswitch.com >>>>> >>>>> -- >>>>> 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 >>>> >>>> >>>> >>>> -- >>>> >>>> Brian West | Co-founder and Developer >>>> >>>> Need Commercial support? email sales at freeswitch.com >>>> >>>> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >>>> >>>> >>>> Email: brian at freeswitch.com >>>> >>>> Mobile: 918-424-9378 >>>> >>>> Website: https://www.FreeSWITCH.com >>>> >>>> [image: https://www.facebook.com/signalwireinc?src=email] >>>> [image: >>>> https://twitter.com/freeswitch] >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Jose Fco. Irles Durá >> > > > -- > Jose Fco. Irles Durá > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Mon Nov 4 19:03:47 2019 From: brian at freeswitch.com (Brian West) Date: Mon, 4 Nov 2019 13:03:47 -0600 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: On Mon, Nov 4, 2019 at 2:20 AM Jose Fco. Irles Durá wrote: > bypass_media=false > Does nothing. > proxy_media=false > Does nothing. > t38_passthru=true (same result setted or not. In sofia profile is setted > the param) > try setting this in vars.xml fax_enable_t38=true (same result setted or not) > Set it as a global in vars.xml sounds like you're having issues with one leg or the other. /b -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From josefu at gmail.com Mon Nov 4 23:05:22 2019 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Tue, 5 Nov 2019 00:05:22 +0100 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: I have tested with t38-passthru=true in the profile and fax_enable_t38=true in the vars.xml and now works. Thanks for the replies. Best regards El lun., 4 nov. 2019 a las 20:04, Brian West () escribió: > > > On Mon, Nov 4, 2019 at 2:20 AM Jose Fco. Irles Durá > wrote: > >> bypass_media=false >> > > Does nothing. > > >> proxy_media=false >> > > Does nothing. > > >> t38_passthru=true (same result setted or not. In sofia profile is setted >> the param) >> > > try setting this in vars.xml > > > fax_enable_t38=true (same result setted or not) >> > > Set it as a global in vars.xml sounds like you're having issues with one > leg or the other. > > /b > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Jose Fco. Irles Durá -------------- next part -------------- An HTML attachment was scrubbed... URL: From s0l6r1s at gmail.com Tue Nov 5 11:56:09 2019 From: s0l6r1s at gmail.com (Max Solaris) Date: Tue, 5 Nov 2019 12:56:09 +0100 Subject: [Freeswitch-users] mod_hash Message-ID: <88787D19-741F-4CFC-BDE4-3D77D66239BA@gmail.com> Hi, I found this module and seems to be very useful, however I have a question. I can see methods supported talks about insert / delete a single key by realm, and this is fine. But, is there a way to delete a complete realm in order to rebuild it? I’m thinking in storing a custom actions LCR in memory and this module fits my needs, but I believe delete a single key each and rebuild a new realm is not so easy do deal with. Thanks Max From spencer.angerbauer at gmail.com Tue Nov 5 05:55:40 2019 From: spencer.angerbauer at gmail.com (Spencer Angerbauer) Date: Mon, 4 Nov 2019 22:55:40 -0700 Subject: [Freeswitch-users] Creating RESTful API xml for dialing outbound numbers into a conference In-Reply-To: <5932E922-84F8-4CA8-A228-F61DB4C9D1FA@gmail.com> References: <5932E922-84F8-4CA8-A228-F61DB4C9D1FA@gmail.com> Message-ID: Hello, I am semi new to FreeSwitch but have worked in Asterisk for many years. I am trying to create a simple originate api call from the webapi module. I have been able to successful create an internal user extension 1001 and connect to 1002 in a default conference room. (See below call format which was successful. http://XX.XX.XX.XX:8080/webapi/originate?user/1001%201002%20XML%20default My question is, How do I take this same command structure to connect multiple external numbers (not internal extensions) into a specific or unique conference room via my SIP trunk line? I have tried multiple variations but am having difficulty finding anything online that shows some good examples of JSON restful webapi’s for originating an outbound call to 2 or more numbers and connecting them into a conference room... I was able to successfully execute the webapi call below for a simple originate api call to an external number and connect but am unsuccessful and bridging these two api commands into what we need. http://XX.XX.XX.XX:8080/webapi/originate?{origination_caller_id_number=18015555555}sofia/carriers/111#180155555551 at XX.XX.XX.XX%20&%20playback(/vr/migstory.wav) Please let me know if you have any resources or API references to simply connect 2+ external calls into a unique conference via the webapi calls. Thank you so much in advance! -Spence -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Tue Nov 5 18:26:55 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Tue, 5 Nov 2019 13:26:55 -0500 Subject: [Freeswitch-users] Creating RESTful API xml for dialing outbound numbers into a conference In-Reply-To: References: <5932E922-84F8-4CA8-A228-F61DB4C9D1FA@gmail.com> Message-ID: Hi Spence -- In your example URL, I would replace the application portion from: &%20playback(/vr/migstory.wav) to %26conference(CONFERENCE_NAME) where the parameter is set to something suitable for that particular instance. Please also note the escaped ampersand sign (%26), be sure to URL-escape all characters which would mislead the browser. On Tue, 5 Nov 2019 at 11:54, Spencer Angerbauer < spencer.angerbauer at gmail.com> wrote: > Hello, > > > I am semi new to FreeSwitch but have worked in Asterisk for many years. I > am trying to create a simple originate api call from the webapi module. I > have been able to successful create an internal user extension 1001 and > connect to 1002 in a default conference room. (See below call format which > was successful. > > http://XX.XX.XX.XX:8080/webapi/originate?user/1001%201002%20XML%20default > > > > My question is, How do I take this same command structure to connect > multiple external numbers (not internal extensions) into a specific or > unique conference room via my SIP trunk line? > > I have tried multiple variations but am having difficulty finding anything > online that shows some good examples of JSON restful webapi’s for > originating an outbound call to 2 or more numbers and connecting them into > a conference room... > > I was able to successfully execute the webapi call below for a simple > originate api call to an external number and connect but am unsuccessful > and bridging these two api commands into what we need. > > > http://XX.XX.XX.XX:8080/webapi/originate?{origination_caller_id_number=18015555555}sofia/carriers/111#180155555551 at XX.XX.XX.XX%20&%20playback(/vr/migstory.wav) > > > Please let me know if you have any resources or API references to simply > connect 2+ external calls into a unique conference via the webapi calls. > > Thank you so much in advance! > > -Spence > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From idokan at gmail.com Wed Nov 6 06:49:03 2019 From: idokan at gmail.com (ik) Date: Wed, 6 Nov 2019 08:49:03 +0200 Subject: [Freeswitch-users] mod_perl on 1.10.1 Message-ID: Hello, I installed under Centos 7 fs 1.10.1 out of the repos - A clean new installation an a clean server. The system replaces an old system from 1.4.x without changing my code, that is written in Perl. Once a week it looks like the loading of the perl script using `perlrun` stop loading the modules properly, getting error that some variables such as %Config were not found. Restarting FS fixes the issue for a week or so. I could not find an open bug on it, but before reporting, I wish to know if there anything else I should be looking for when such issue found? Thank you Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Wed Nov 6 18:32:07 2019 From: brian at freeswitch.com (Brian West) Date: Wed, 6 Nov 2019 12:32:07 -0600 Subject: [Freeswitch-users] Fax T38 problems in proxy media In-Reply-To: References: Message-ID: Excellent! On Mon, Nov 4, 2019 at 5:06 PM Jose Fco. Irles Durá wrote: > I have tested with t38-passthru=true in the profile > and fax_enable_t38=true in the vars.xml and now works. > > Thanks for the replies. > > Best regards > > El lun., 4 nov. 2019 a las 20:04, Brian West () > escribió: > >> >> >> On Mon, Nov 4, 2019 at 2:20 AM Jose Fco. Irles Durá >> wrote: >> >>> bypass_media=false >>> >> >> Does nothing. >> >> >>> proxy_media=false >>> >> >> Does nothing. >> >> >>> t38_passthru=true (same result setted or not. In sofia profile is setted >>> the param) >>> >> >> try setting this in vars.xml >> >> >> fax_enable_t38=true (same result setted or not) >>> >> >> Set it as a global in vars.xml sounds like you're having issues with one >> leg or the other. >> >> /b >> >> >> -- >> >> Brian West | Co-founder and Developer >> >> Need Commercial support? email sales at freeswitch.com >> >> FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 >> >> >> Email: brian at freeswitch.com >> >> Mobile: 918-424-9378 >> >> Website: https://www.FreeSWITCH.com >> >> [image: https://www.facebook.com/signalwireinc?src=email] >> [image: >> https://twitter.com/freeswitch] >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Jose Fco. Irles Durá > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From smiller at kc1awv.net Wed Nov 6 14:51:05 2019 From: smiller at kc1awv.net (Stephen Miller) Date: Wed, 6 Nov 2019 09:51:05 -0500 Subject: [Freeswitch-users] fs_rpt.pl radio interface and mod_echolink Message-ID: Hello everyone! I am wondering if there still exists a copy of fs_rpt.pl and mod_echolink for FreeSWITCH. I have a few applications that I would like to explore... and really eventually switch off from Asterisk. Any help would be greatly appreciated! Thank you! -- Steve Miller KC1AWV -------------- next part -------------- An HTML attachment was scrubbed... URL: From akndoye at groupechaka.com Thu Nov 7 13:34:49 2019 From: akndoye at groupechaka.com (Abdou khadre NDOYE) Date: Thu, 7 Nov 2019 13:34:49 +0000 Subject: [Freeswitch-users] Local stream Message-ID: <16367039-4530-4A69-989B-E8D22036A9FF@groupechaka.com> Hi, I have some trouble using mod_local_stream, When I place a file in the right path configured in local_stream.conf.xml file doesn’t played. Sometime I have this error when I use rate 8000 : 2019-11-07 10:53:14.620285 [DEBUG] switch_core_media.c:8494 Audio params are unchanged for sofia/internal/338600485 at 172.16.10.5. 2019-11-07 10:53:14.660286 [DEBUG] switch_ivr.c:625 sofia/internal/315 at 172.16.10.50:5060 Command Execute playback(local_stream://moh) EXECUTE sofia/internal/315 at 172.16.10.50:5060 playback(local_stream://moh) 2019-11-07 10:53:14.660286 [DEBUG] switch_core_file.c:389 File moh sample rate 8000 doesn't match requested rate 16000 2019-11-07 10:53:14.660286 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16 at 16000hz 1 channels 20ms 2019-11-07 10:53:14.680286 [NOTICE] switch_core_media.c:15605 Deactivating write resampler My config ; I need help. Thank From mike at freeswitch.org Thu Nov 7 18:16:00 2019 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 7 Nov 2019 11:16:00 -0700 Subject: [Freeswitch-users] fs_rpt.pl radio interface and mod_echolink In-Reply-To: References: Message-ID: <0E4571CF-26F0-421D-922D-2D859FFA3231@freeswitch.org> https://freeswitch.org/confluence/display/FREESWITCH/fs_rpt.pl I dont think echo link stuff was ever completed and cant recall who worked on that. > On Nov 6, 2019, at 7:51 AM, Stephen Miller wrote: > > Hello everyone! > > I am wondering if there still exists a copy of fs_rpt.pl and mod_echolink for FreeSWITCH. I have a few applications that I would like to explore... and really eventually switch off from Asterisk. > > Any help would be greatly appreciated! > > Thank you! > > -- > Steve Miller > KC1AWV -------------- next part -------------- An HTML attachment was scrubbed... URL: From david at net-work.net Fri Nov 8 12:47:16 2019 From: david at net-work.net (David Dean) Date: Fri, 8 Nov 2019 12:47:16 +0000 Subject: [Freeswitch-users] ERR has no write codec. [INCOMPATIBLE_DESTINATION] Message-ID: Hi I wonder if anybody can help. We are getting an error creeping in on a production system. It is happening randomly which of course makes it difficult to track down. A call to a SIP registered endpoint (VoIP handset) randomly fails, it rings, then when it is answered the call fails. It could be to any of the end points on our system, it doesn't happen every call, in fact it only happens on prepositionally very few calls. If we restart the Freeswitch service the frequency of calls effected reduces to nothing and then builds up again. Logs show (I have replaced our IP address with {IP ADDRESS}) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.857442 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [ERR] switch_core_media.c:15418 sofia/internal/5033305@{IP ADDRESS}:38532 has no write codec. freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [NOTICE] switch_core_media.c:15419 Hangup sofia/internal/5033305@{IP ADDRESS}:38532 [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/5033305@{IP ADDRESS}:38532 skip receive message [UNBRIDGE] (channel is hungup already) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.837454 [DEBUG] sofia.c:7301 Remote SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=5033305 38480 137 IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=Mapping freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 38480 RTP/AVP 0 101 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Local SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=FreeSWITCH 1573174674 1573174675 IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=FreeSWITCH freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 32570 RTP/AVP 0 101 13 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:13 CN/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=sendrecv freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Any ideas would be appreciated. David J Dean From ka.rathod99 at gmail.com Fri Nov 8 16:10:37 2019 From: ka.rathod99 at gmail.com (Abhilash Rathod) Date: Fri, 8 Nov 2019 21:40:37 +0530 Subject: [Freeswitch-users] Audio delay on the conference calls Message-ID: Hello, I am using FreeSWITCH version 1.8.7 on Centos 7. I am facing an audio delay on the conference call. Three users are on the conference call. In the initial time, audio is fine, there was no audio delay but audio delay increasing wisely as call time duration increasing. Below I have mentioned some delay examples which I have faced. Call Time: Delay(approx): 00:45 1.02s 02:30 1.10s 04:35 1.25s 07:45 1.60s 10:35 1.80s 15:15 2.60s 21:40 3.60s 27:10 4.25s 30:00 4.30s I have applied two changes on the FreeSWITCH for the fix this issue, but still, i am facing audio delay issue. 1. set rtp_time_name= none in dialplan 2. set in sofia settings. Is there any other solution available to resolve this issue. Advance Thanks for the help. -- *Thanks & Regards* *Abhilash Rathod* *[Skype]:- rathod.abhilash* *[Whats App]:- **+91 9016232506* -------------- next part -------------- An HTML attachment was scrubbed... URL: From hyavari at rocketmail.com Sun Nov 10 10:32:53 2019 From: hyavari at rocketmail.com (H Yavari) Date: Sun, 10 Nov 2019 10:32:53 +0000 (UTC) Subject: [Freeswitch-users] ERR has no write codec. [INCOMPATIBLE_DESTINATION] In-Reply-To: References: Message-ID: <1849171270.2132948.1573381973452@mail.yahoo.com> Hi David, Have u checked your codecs in FS and end points? as we can see, you have some mismatch codec or transcoding problem. Regards,HYavari On Sunday, November 10, 2019, 1:51:47 PM GMT+3:30, David Dean wrote: Hi I wonder if anybody can help. We are getting an error creeping in on a production system. It is happening randomly which of course makes it difficult to track down. A call to a SIP registered endpoint (VoIP handset) randomly fails, it rings, then when it is answered the call fails. It could be to any of the end points on our system, it doesn't happen every call, in fact it only happens on prepositionally very few calls. If we restart the Freeswitch service the frequency of calls effected reduces to nothing and then builds up again. Logs show (I have replaced our IP address with {IP ADDRESS}) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.857442 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [ERR] switch_core_media.c:15418 sofia/internal/5033305@{IP ADDRESS}:38532 has no write codec. freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [NOTICE] switch_core_media.c:15419 Hangup sofia/internal/5033305@{IP ADDRESS}:38532 [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/5033305@{IP ADDRESS}:38532 skip receive message [UNBRIDGE] (channel is hungup already) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.837454 [DEBUG] sofia.c:7301 Remote SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=5033305 38480 137 IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=Mapping freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 38480 RTP/AVP 0 101 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Local SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=FreeSWITCH 1573174674 1573174675 IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=FreeSWITCH freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 32570 RTP/AVP 0 101 13 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:13 CN/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=sendrecv freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Any ideas would be appreciated. David J Dean _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users 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 Sun Nov 10 12:28:18 2019 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sun, 10 Nov 2019 13:28:18 +0100 Subject: [Freeswitch-users] ERR has no write codec. [INCOMPATIBLE_DESTINATION] In-Reply-To: References: Message-ID: Hi, Can you try disabling the comfort noise generation? You can do it by setting the profile param suppress-cng or the channel variable suppress_cng to true. It should remove the "a=rtpmap:13 CN/8000" line from your local sdp. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sip.solutions linkedin.com/in/vallimamod . > On 8 Nov 2019, at 13:47, David Dean wrote: > > Hi > > I wonder if anybody can help. We are getting an error creeping in on a production system. It is happening randomly which of course makes it difficult to track down. > > A call to a SIP registered endpoint (VoIP handset) randomly fails, it rings, then when it is answered the call fails. It could be to any of the end points on our system, it doesn't happen every call, in fact it only happens on prepositionally very few calls. If we restart the Freeswitch service the frequency of calls effected reduces to nothing and then builds up again. > > Logs show (I have replaced our IP address with {IP ADDRESS}) > > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.857442 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [ERR] switch_core_media.c:15418 sofia/internal/5033305@{IP ADDRESS}:38532 has no write codec. > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [NOTICE] switch_core_media.c:15419 Hangup sofia/internal/5033305@{IP ADDRESS}:38532 [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/5033305@{IP ADDRESS}:38532 skip receive message [UNBRIDGE] (channel is hungup already) > > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.837454 [DEBUG] sofia.c:7301 Remote SDP: > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=5033305 38480 137 IN IP4 51.148.123.105 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=Mapping > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 51.148.123.105 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 38480 RTP/AVP 0 101 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 > > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Local SDP: > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=FreeSWITCH 1573174674 1573174675 IN IP4 178.128.173.117 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=FreeSWITCH > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 178.128.173.117 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 32570 RTP/AVP 0 101 13 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:13 CN/8000 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=sendrecv > freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 > > Any ideas would be appreciated. > > David J Dean > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 nathan at robotics.net Mon Nov 11 13:55:05 2019 From: nathan at robotics.net (Nathan Stratton) Date: Mon, 11 Nov 2019 08:55:05 -0500 Subject: [Freeswitch-users] Video / Audio Voicemail Message-ID: I loaded mp4v2, but get the following error when I leave a voicemail. 2019-11-11 13:51:30.884778 [DEBUG] mod_mp4v2.c:357 sample rate: 44100, channels: 1 2019-11-11 13:51:30.884778 [INFO] mod_mp4v2.c:402 Opening File [/tmp/2ca88130-5f57-4728-9703-c57bbfb365fe.mp4] 44100hz 2019-11-11 13:51:30.884778 [WARNING] switch_core_codec.c:727 Codec PCMU Exists but not at the desired implementation. 44100hz 20ms 1ch 2019-11-11 13:51:30.884778 [ERR] mod_mp4v2.c:415 Audio Codec Activation Fail Any ideas? ><> nathan stratton -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Nov 11 14:55:50 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 11 Nov 2019 15:55:50 +0100 Subject: [Freeswitch-users] Video / Audio Voicemail In-Reply-To: References: Message-ID: Iirc, use mod_av ( not mp4v2) On Mon, Nov 11, 2019, 15:34 Nathan Stratton wrote: > I loaded mp4v2, but get the following error when I leave a voicemail. > > 2019-11-11 13:51:30.884778 [DEBUG] mod_mp4v2.c:357 sample rate: 44100, > channels: 1 > 2019-11-11 13:51:30.884778 [INFO] mod_mp4v2.c:402 Opening File > [/tmp/2ca88130-5f57-4728-9703-c57bbfb365fe.mp4] 44100hz > 2019-11-11 13:51:30.884778 [WARNING] switch_core_codec.c:727 Codec PCMU > Exists but not at the desired implementation. 44100hz 20ms 1ch > 2019-11-11 13:51:30.884778 [ERR] mod_mp4v2.c:415 Audio Codec Activation > Fail > > Any ideas? > > ><> > nathan stratton > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Mon Nov 11 08:25:23 2019 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Mon, 11 Nov 2019 09:25:23 +0100 Subject: [Freeswitch-users] Freeswitch 1.10.X mod_limit db configuration Message-ID: Hi I need some help from community, I have a question regarding mod_limit db configuration in modern freeswitch release where we have new db drivers. I tried to use following construction: When I try to load mod_db I`m getting following error: 2019-11-11 09:22:11.813986 [ERR] switch_xml_config.c:267 Invalid value [mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;] for parameter [odbc-dsn] freeswitch at loopkam> Item name: [odbc-dsn] Type: string (optional) Syntax: dsn:username:password Help: If set, the ODBC DSN used by the limit and db applications I have following version installed: FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready Thanks for advices! -- Best regards, Volodymyr From saqibnawaz5812 at gmail.com Mon Nov 11 19:13:06 2019 From: saqibnawaz5812 at gmail.com (Saqib Nawaz) Date: Tue, 12 Nov 2019 00:13:06 +0500 Subject: [Freeswitch-users] Failed DTMF sanity check Message-ID: Hi Guys, I'm getting this error message printed on my console [ERR] switch_rtp.c:2013 Failed DTMF sanity check. Why this is popping up ? how can I prevent this from occurring? Regards, Saqib Nawaz. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Mon Nov 11 19:40:14 2019 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 11 Nov 2019 11:40:14 -0800 Subject: [Freeswitch-users] Audio delay on the conference calls In-Reply-To: References: Message-ID: <7A3722C6-AE5D-4ED6-88F0-CB8475D2D30D@freeswitch.org> Did you try the current release? > On Nov 8, 2019, at 8:10 AM, Abhilash Rathod wrote: > > Hello, > > I am using FreeSWITCH version 1.8.7 on Centos 7. I am facing an audio delay on the conference call. Three users are on the conference call. > In the initial time, audio is fine, there was no audio delay but audio delay increasing wisely as call time duration increasing. Below I have mentioned some delay examples which I have faced. > > Call Time: Delay(approx): > 00:45 1.02s > 02:30 1.10s > 04:35 1.25s > 07:45 1.60s > 10:35 1.80s > 15:15 2.60s > 21:40 3.60s > 27:10 4.25s > 30:00 4.30s > > I have applied two changes on the FreeSWITCH for the fix this issue, but still, i am facing audio delay issue. > 1. set rtp_time_name= none in dialplan > 2. set in sofia settings. > > Is there any other solution available to resolve this issue. > Advance Thanks for the help. From ciprian.dosoftei at gmail.com Mon Nov 11 19:51:40 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Mon, 11 Nov 2019 14:51:40 -0500 Subject: [Freeswitch-users] Failed DTMF sanity check In-Reply-To: References: Message-ID: It typically means the device on the other end of the channel is sending in malformed RFC2833 DTMF. What FreeSWITCH version are you using, and also, do you know what kind of device is connected on the other end? On Mon, 11 Nov 2019 at 14:28, Saqib Nawaz wrote: > Hi Guys, > > I'm getting this error message printed on my console > > [ERR] switch_rtp.c:2013 Failed DTMF sanity check. > > Why this is popping up ? how can I prevent this from occurring? > > Regards, > Saqib Nawaz. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From andywolk at gmail.com Mon Nov 11 19:59:03 2019 From: andywolk at gmail.com (Andrey Wolk) Date: Mon, 11 Nov 2019 23:59:03 +0400 Subject: [Freeswitch-users] Freeswitch 1.10.X mod_limit db configuration Message-ID: Since it's a regex "^pgsql|^odbc|^sqlite|[^:]+:[^:]*:.*" "dsn:username:password" Try adding : at the end of the value. > From: Volodymyr Fedorov > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Mon, 11 Nov 2019 09:25:23 +0100 > Subject: [Freeswitch-users] Freeswitch 1.10.X mod_limit db configuration > Hi I need some help from community, I have a question regarding > mod_limit db configuration in modern freeswitch release where we have > new db drivers. > I tried to use following construction: > > > > value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;"/> > > > > When I try to load mod_db I`m getting following error: > 2019-11-11 09:22:11.813986 [ERR] switch_xml_config.c:267 Invalid value > > [mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;] > for parameter [odbc-dsn] > freeswitch at loopkam> > Item name: [odbc-dsn] > Type: string (optional) > Syntax: dsn:username:password > Help: If set, the ODBC DSN used by the limit and db applications > > I have following version installed: > FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready > > Thanks for advices! > -- > Best regards, > Volodymyr > -------------- next part -------------- An HTML attachment was scrubbed... URL: From saqibnawaz5812 at gmail.com Mon Nov 11 21:13:02 2019 From: saqibnawaz5812 at gmail.com (Saqib Nawaz) Date: Tue, 12 Nov 2019 02:13:02 +0500 Subject: [Freeswitch-users] Failed DTMF sanity check In-Reply-To: References: Message-ID: Thanks for your reply, I'm using FreeSWITCH version: 1.6.9. No idea of the device connected to other ends. On Tue, 12 Nov 2019 at 01:26, Ciprian Dosoftei wrote: > It typically means the device on the other end of the channel is sending > in malformed RFC2833 DTMF. > > What FreeSWITCH version are you using, and also, do you know what kind of > device is connected on the other end? > > On Mon, 11 Nov 2019 at 14:28, Saqib Nawaz > wrote: > >> Hi Guys, >> >> I'm getting this error message printed on my console >> >> [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >> >> Why this is popping up ? how can I prevent this from occurring? >> >> Regards, >> Saqib Nawaz. >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From info at behrend-cs.de Mon Nov 11 22:21:05 2019 From: info at behrend-cs.de (Walter Behrend) Date: Mon, 11 Nov 2019 23:21:05 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster Message-ID: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> Hello there, hope someone else also had the problem - and found a solution for it. My "internal" profile has TLS enabled with tlsv1, 1.1 and 1.2 - this worked like a charm on stretch. I'm using the freeswitch-repos. I upgraded to buster and here my problems started. Seems the gentls_cert only creates SHA1 (CA)Certificates - so freeswitch started with openssl error messages "md too weak". Tried at first to bypass this error by setting the tls_ciphers to "DEFAULT:@SECLEVEL=0" but this error still occured. So as a consequence, I modified the gentls_cert script and replaced everywhere the parameter -sha1 with -sha256. This error disappeared now, but the next one is coming up. It seems it does not matter which value I set for "tls_version" - in every case, my TLS enabled port only accepts TLS 1.3 connections. I have the problem that we're also using older phones which only support TLS 1.0. Error message is: tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol I tried with openssl s_client and the parameters -tls1 -tls1_1 and so on - it really only worked for -tls1_3 Any idea about this? setting tls_version to tlsv1,tlsv1.1,tlsv1.2 does not change anything. Also setting the value just to tlsv1 does not help, I verified this with the phones AND with openssl s_client. Still only TLS 1.3 gives results here. Thanks in advance... BR Walter -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Tue Nov 12 14:27:02 2019 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Tue, 12 Nov 2019 14:27:02 +0000 Subject: [Freeswitch-users] dotnet core 3.0 vs. mod_managed Message-ID: <0ead7525996c45d2bbba232d80ccbd0f@c4b.de> Hi, I try to use dotnet core 3 apps with mod_managed on a Debian 10 machine. I’m the first who try this or have anyone some experience with this issue? If I have results, I’ll get it back to the community. Nice Regards Alex -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathan at robotics.net Tue Nov 12 16:08:25 2019 From: nathan at robotics.net (Nathan Stratton) Date: Tue, 12 Nov 2019 11:08:25 -0500 Subject: [Freeswitch-users] Video / Audio Voicemail In-Reply-To: References: Message-ID: On Mon, Nov 11, 2019 at 9:56 AM Giovanni Maruzzelli wrote: > Iirc, use mod_av ( not mp4v2) > Sir, you rock! I could not find that anywhere! -Nathan -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Tue Nov 12 20:27:16 2019 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Tue, 12 Nov 2019 21:27:16 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster In-Reply-To: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> Message-ID: <20191112202715.GA3584@darth.lan> On Mon, Nov 11, 2019 at 11:21:05PM +0100, Walter Behrend wrote: > Hello there, Hi Walter, > I upgraded to buster and here my problems started. Seems the gentls_cert > only creates SHA1 (CA)Certificates - so freeswitch started with openssl > error messages "md too weak". Tried at first to bypass this error by setting > the tls_ciphers to "DEFAULT:@SECLEVEL=0" but this error still occured. I don't have Debian, I got OpenWrt with openssl 1.1.1d and FS 1.10.1. Updated /etc/ssl/openssl.cnf like this: https://sources.debian.org/src/openssl/1.1.1c-1/debian/patches/Set-systemwide-default-settings-for-libssl-users.patch/ I also used gentls_cert to create CA & server cert. But I don't get "md too weak" when starting FS. > So as a consequence, I modified the gentls_cert script and replaced > everywhere the parameter -sha1 with -sha256. This error disappeared now, but > the next one is coming up. > > It seems it does not matter which value I set for "tls_version" - in every > case, my TLS enabled port only accepts TLS 1.3 connections. I have the > problem that we're also using older phones which only support TLS 1.0. > > Error message is: > > tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL > routines:tls_early_post_process_client_hello:unsupported protocol Works fine here: 4 0.004770 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.193706 192.168.0.1 → 192.168.0.120 TLSv1.2 1514 Server Hello 7 0.193809 192.168.0.1 → 192.168.0.120 TLSv1.2 871 Certificate, Server Key Exchange, Server Hello Done 10 0.256056 192.168.0.120 → 192.168.0.1 TLSv1.2 141 Client Key Exchange 12 0.269076 192.168.0.120 → 192.168.0.1 TLSv1.2 72 Change Cipher Spec 14 0.269344 192.168.0.120 → 192.168.0.1 TLSv1.2 103 Encrypted Handshake Message With openssl s_client I can also connect with TLS1.0, 1.1, 1.2 and 1.3 (which suggests that FS isn't using system openssl config). > Any idea about this? setting tls_version to tlsv1,tlsv1.1,tlsv1.2 does > not change anything. Also setting the value just to tlsv1 does not > help, I verified this with the phones AND with openssl s_client. Still > only TLS 1.3 gives results here. The only way to reproduce your result was to set tls-version to tls1_3. When you grep your FS log for tls-version, do you see "tlsv1,tlsv1.1,tlsv1.2" or "tlsv1_3"? Regards, Seb From sebastian_ml at gmx.net Tue Nov 12 21:14:54 2019 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Tue, 12 Nov 2019 22:14:54 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster In-Reply-To: <20191112202715.GA3584@darth.lan> References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> <20191112202715.GA3584@darth.lan> Message-ID: <20191112211454.GA3609@darth.lan> On Tue, Nov 12, 2019 at 09:27:16PM +0100, Sebastian Kemper wrote: > On Mon, Nov 11, 2019 at 11:21:05PM +0100, Walter Behrend wrote: > > It seems it does not matter which value I set for "tls_version" - in every > > case, my TLS enabled port only accepts TLS 1.3 connections. I have the > > problem that we're also using older phones which only support TLS 1.0. > > > > Error message is: > > > > tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL > > routines:tls_early_post_process_client_hello:unsupported protocol > > Works fine here: > > 4 0.004770 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello > 6 0.193706 192.168.0.1 → 192.168.0.120 TLSv1.2 1514 Server Hello > 7 0.193809 192.168.0.1 → 192.168.0.120 TLSv1.2 871 Certificate, Server Key Exchange, Server Hello Done > 10 0.256056 192.168.0.120 → 192.168.0.1 TLSv1.2 141 Client Key Exchange > 12 0.269076 192.168.0.120 → 192.168.0.1 TLSv1.2 72 Change Cipher Spec > 14 0.269344 192.168.0.120 → 192.168.0.1 TLSv1.2 103 Encrypted Handshake Message > > With openssl s_client I can also connect with TLS1.0, 1.1, 1.2 and 1.3 > (which suggests that FS isn't using system openssl config). Actually it doesn't work fine here after all. I had updated /etc/ssl/openssl.cnf with the Debian changes, but actually openssl_conf = default_conf was overwritten by some OpenWrt config snippet later. I amended that and now when I set tls_version to something specific I get 4 0.010062 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010927 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Protocol Version) and when I leave it unset I get 4 0.009440 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010884 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Handshake Failure) Regards, Seb From cong.wang.itsherpa at gmail.com Wed Nov 13 02:49:07 2019 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Wed, 13 Nov 2019 11:49:07 +0900 Subject: [Freeswitch-users] L16 Codec in mod_conference Message-ID: Hey all, I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: session:execute("pre_answer") session:execute("conference_set_auto_outcall", "user/" .. args.call_user) session:execute("conference", "testroom at default") My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: opus But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 variable_rtp_use_codec_name: opus variable_rtp_use_codec_fmtp: useinbandfec%3D1 variable_rtp_use_codec_rate: 48000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c variable_original_read_codec: opus variable_write_codec: opus variable_read_codec: L16 Both tests are based on Linphone offical app. During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. Any suggestion would be appreciated. Regards, C.Wang -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Wed Nov 13 14:04:13 2019 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Wed, 13 Nov 2019 15:04:13 +0100 Subject: [Freeswitch-users] Freeswitch 1.10.X mod_limit db configuration In-Reply-To: References: Message-ID: Hi Andrey, Thank you for given example. Works like a charm now! BR, Vova On Mon, Nov 11, 2019 at 9:33 PM Andrey Wolk wrote: > > Since it's a regex > "^pgsql|^odbc|^sqlite|[^:]+:[^:]*:.*" > "dsn:username:password" > > Try adding : at the end of the value. > > value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;:"/> > >> >> From: Volodymyr Fedorov >> To: FreeSWITCH Users Help >> Cc: >> Bcc: >> Date: Mon, 11 Nov 2019 09:25:23 +0100 >> Subject: [Freeswitch-users] Freeswitch 1.10.X mod_limit db configuration >> Hi I need some help from community, I have a question regarding >> mod_limit db configuration in modern freeswitch release where we have >> new db drivers. >> I tried to use following construction: >> >> >> > value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;"/> >> >> >> >> When I try to load mod_db I`m getting following error: >> 2019-11-11 09:22:11.813986 [ERR] switch_xml_config.c:267 Invalid value >> [mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;] >> for parameter [odbc-dsn] >> freeswitch at loopkam> >> Item name: [odbc-dsn] >> Type: string (optional) >> Syntax: dsn:username:password >> Help: If set, the ODBC DSN used by the limit and db applications >> >> I have following version installed: >> FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready >> >> Thanks for advices! >> -- >> Best regards, >> Volodymyr > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best regards, Volodymyr From asilva at wirelessmundi.com Wed Nov 13 15:53:25 2019 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Wed, 13 Nov 2019 16:53:25 +0100 Subject: [Freeswitch-users] FreeSwitch Hangs and Fails to Process anything In-Reply-To: References: Message-ID: <23a1fa65-bf3a-896f-c29b-bb1b5319787f@wirelessmundi.com> It happen to me also using tag v1.10 and current master, using debian 10.  In version 1.8.7 is not happing with debian jessie. A just submit a git issue where i got a gcore from the freezing process, i logs there is nothing wrong. Link to issue: https://github.com/signalwire/freeswitch/issues/125 On 23/10/2019 22:52, Guillermo Ruiz Camauer wrote: > From https://freeswitch.org/confluence/display/FREESWITCH/Debugger > Please read the rest of that page and also look at > https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA : > > > > Recompiling with debug symbols on > > /Note: FreeSWITCH is compiled with debug symbols on by default !/ > > export CFLAGS="-g -ggdb" > export MOD_CFLAGS="-g -ggdb" > ./configure > > > Creating core files > > For core files to be created when a crash occurs, set the core ulimit > to unlimited before starting FreeSWITCH > > $ ulimit -c unlimited > $ ./freeswitch > > The core file should be located in the directory where FreeSWITCH was > started (i.e., if you were in the /tmp directory when you typed the > command to start FreeSWITCH, then there should be a file called > something like /tmp/core.xxx). > > NOTE: On OS X, core files are dumped to a hidden directory called > /cores by default, not the current directory! > > > On Wed, Oct 23, 2019 at 5:05 PM Lloyd Aloysius > > wrote: > > I never have this issues. I was using version 1.4 version before. > Now upgrade to 1.10... See this problem. > > How to compile with symbols? > > On Wed, Oct 23, 2019 at 12:23 PM David Villasmil > > wrote: > > Hello, > > Why don't you compile with symbols and when it hangs get a > backtrace, then file an issue on github. > This is important for all, if there's a but somewhere, it > should be found and fixed. > > But i have many boxes (debian 8/9, version 1.6.20/1.6.18) > doing 60-100 channels and never seen anything like that. > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > > On Wed, Oct 23, 2019 at 3:43 PM Lloyd Aloysius > > > wrote: > > Regular PBX calls between users > > On Wed, Oct 23, 2019 at 9:11 AM David Villasmil > > wrote: > > Why is it freezing? Anything in the logs? That’s not > normal at all. > What are those 20-30 calls doing? > > On Wed, 23 Oct 2019 at 13:18, Lloyd Aloysius > > wrote: > > Finally I setup a cronjob to restart freeswitch > everyday. Otherwise it hang every 2 days. Does > this is a know problem to the most recent version. > I even try the master.Same issue. > > Hardware - 16 Core CPU, 96GB Memeory, SSDs. > > Average calls volume is 20~30 calls > > Does the following startup script is valid. May be > I am missing some requirements for version 1.10 ? > > > ; This file in installations built from Master can > be found in > ; /usr/src/freeswitch.git/debian > ; or > ; /usr/src/freeswitch/debian > [Unit] > Description=freeswitch > After=syslog.target network.target local-fs.target > mysql.service apache2.service > > [Service] > ; service > Type=forking > PIDFile=/usr/local/freeswitch/run/freeswitch.pid > PermissionsStartOnly=true > ; blank ExecStart= line flushes the list > ExecStart= > ExecStart=/usr/local/freeswitch/bin/freeswitch -u > freeswitch -g freeswitch -ncwait -nonat -rp > TimeoutSec=45s > Restart=always > ; exec > WorkingDirectory=/usr/local/freeswitch/bin > User=root > Group=daemon > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > LimitSTACK=250000 > LimitRTPRIO=infinity > LimitRTTIME=infinity > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > WantedBy=multi-user.target > > > > > On Mon, Oct 14, 2019 at 12:49 PM Lloyd Aloysius > > wrote: > > it happened again. Create a issue and attached > the back trace. > > > > > > On Sun, Oct 13, 2019 at 1:44 PM John Covici > > wrote: > > Why don't you compile with symbols and > when it hangs get a backtrace, > then file an issue on github. > > On Sun, 13 Oct 2019 11:07:13 -0400, > David Villasmil wrote: > > > > [1  ] > > [1.1  (quoted-printable)>] > > [1.2  ] > > Your fail2ban will only block IPs trying > to register unsuccessfully. > > > > What happens with those calls? Do they > get dropped? Where’s the logging on those > calls? Sled exits “console log level debug” > > > > On Sun, 13 Oct 2019 at 15:59, Lloyd > Aloysius > wrote: > > > >  I do not see any errors... only see > these entries. The following in my > fail2ban... Does this correct > > > >  [Definition] > > > >  failregex = ^\.\d+ \[WARNING\] > sofia_reg\.c:\d+ SIP auth > (failure|challenge) \((REGISTER|INVITE)\) > on sofia profile \'[^']+\' for \[.*\] from > ip $ > >              ^\.\d+ \[WARNING\] > sofia_reg\.c:\d+ Can't find user > \[\d+@\d+\.\d+\.\d+\.\d+\] from $ > > > >  ignoreregex = > > > >  On Sun, Oct 13, 2019 at 10:16 AM David > Villasmil > > wrote: > > > >  Is fail2ban blocking those? Doesn’t > seem like it is. You should probably > double-check it’s config. > > > >  In any case, there’s nothing in the log > you pasted showing any errors. Please > paste the errors (if any) > > > >  On Sun, 13 Oct 2019 at 13:59, Lloyd > Aloysius > wrote: > > > >  Hangs again. I was using version 1.4 > before. Just updated to FreeSWITCH > (Version 1.10.1-release git f999022 > 2019-08-20 16:54:04Z 64bit) is ready > > > >  Please find the log below. Looks like > someone try to send many calls and switch > hang. I have Fail2Ban setup for this > machine. Machine is Very powerful > > > >  HP 360 Gen 8 Server 16 Cores CPU, 96GB > Memeory , SSD Drives. > > > > 24719e77-462f-4c49-b7e4-3e3ca00e5d59 > 2019-10-13 07:01:15.851407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/101 at A.B.C.D > [24719e77-462f-4c49-b7e4-3e3ca00e5d59] > > 24719e77-462f-4c49-b7e4-3e3ca00e5d59 > 2019-10-13 07:01:15.851407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/101 at A.B.C.D) Running > State Change CS_NEW (Cur 2 Tot 1875) > > a4281704-fb77-4fe6-8598-8ca506ecb7e8 > 2019-10-13 07:01:37.111410 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [a4281704-fb77-4fe6-8598-8ca506ecb7e8] > > a4281704-fb77-4fe6-8598-8ca506ecb7e8 > 2019-10-13 07:01:37.111410 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 3 Tot 1876) > > ff2f92d8-9d1c-4878-bb23-2ddac61bce40 > 2019-10-13 07:02:30.971414 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/101 at A.B.C.D > [ff2f92d8-9d1c-4878-bb23-2ddac61bce40] > > ff2f92d8-9d1c-4878-bb23-2ddac61bce40 > 2019-10-13 07:02:30.971414 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/101 at A.B.C.D) Running > State Change CS_NEW (Cur 4 Tot 1877) > > 0e2a1e57-65fa-490e-9368-eeaa92de79d1 > 2019-10-13 07:02:57.791407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [0e2a1e57-65fa-490e-9368-eeaa92de79d1] > > 0e2a1e57-65fa-490e-9368-eeaa92de79d1 > 2019-10-13 07:02:57.791407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 5 Tot 1878) > > 614b6aa3-9c6b-4377-8f95-a2aa0ebe603d > 2019-10-13 07:03:46.331406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/101 at A.B.C.D > [614b6aa3-9c6b-4377-8f95-a2aa0ebe603d] > > 614b6aa3-9c6b-4377-8f95-a2aa0ebe603d > 2019-10-13 07:03:46.331406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/101 at A.B.C.D) Running > State Change CS_NEW (Cur 6 Tot 1879) > > a650801a-ade3-4190-b920-6f79dba49d3a > 2019-10-13 07:04:20.771417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [a650801a-ade3-4190-b920-6f79dba49d3a] > > a650801a-ade3-4190-b920-6f79dba49d3a > 2019-10-13 07:04:20.771417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 7 Tot 1880) > > 883ebcfb-1ff5-4414-bd13-b06d7e00168d > 2019-10-13 07:05:41.211416 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [883ebcfb-1ff5-4414-bd13-b06d7e00168d] > > 883ebcfb-1ff5-4414-bd13-b06d7e00168d > 2019-10-13 07:05:41.211416 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 8 Tot 1881) > > 376737a5-904b-49f7-a9ae-0a446806edca > 2019-10-13 07:07:03.331417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [376737a5-904b-49f7-a9ae-0a446806edca] > > 376737a5-904b-49f7-a9ae-0a446806edca > 2019-10-13 07:07:03.331417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 9 Tot 1882) > > 7df169e7-ea7f-4f80-8e0b-6569fab356b5 > 2019-10-13 07:08:24.411412 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [7df169e7-ea7f-4f80-8e0b-6569fab356b5] > > 7df169e7-ea7f-4f80-8e0b-6569fab356b5 > 2019-10-13 07:08:24.411412 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 10 Tot 1883) > > fa96fd08-99a1-436f-9b43-a5f583f8a0c2 > 2019-10-13 07:09:45.671417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [fa96fd08-99a1-436f-9b43-a5f583f8a0c2] > > fa96fd08-99a1-436f-9b43-a5f583f8a0c2 > 2019-10-13 07:09:45.671417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 11 Tot 1884) > >  2019-10-13 07:10:01.751407 [WARNING] > sofia.c:2298 MSG Thread 5 Started > >  2019-10-13 07:10:01.751407 [WARNING] > sofia.c:2298 MSG Thread 4 Started > > 5ebba99b-9f81-4948-9a2d-eedcd9b32abb > 2019-10-13 07:11:06.911408 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [5ebba99b-9f81-4948-9a2d-eedcd9b32abb] > > 5ebba99b-9f81-4948-9a2d-eedcd9b32abb > 2019-10-13 07:11:06.911408 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 12 Tot 1885) > > 755828aa-8838-4c38-a1c8-6f0ba82da14d > 2019-10-13 07:12:29.071406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [755828aa-8838-4c38-a1c8-6f0ba82da14d] > > 755828aa-8838-4c38-a1c8-6f0ba82da14d > 2019-10-13 07:12:29.071406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 13 Tot 1886) > > f290f9f9-6ce2-495d-895e-d9f9be0276e8 > 2019-10-13 07:13:49.291415 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [f290f9f9-6ce2-495d-895e-d9f9be0276e8] > > f290f9f9-6ce2-495d-895e-d9f9be0276e8 > 2019-10-13 07:13:49.291415 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 14 Tot 1887) > >  2019-10-13 07:14:22.231392 [WARNING] > sofia.c:2298 MSG Thread 7 Started > >  2019-10-13 07:14:22.231392 [WARNING] > sofia.c:2298 MSG Thread 6 Started > > 4d6127f4-852b-430a-9c6a-ed0fdcfae0c1 > 2019-10-13 07:15:09.311406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [4d6127f4-852b-430a-9c6a-ed0fdcfae0c1] > > 4d6127f4-852b-430a-9c6a-ed0fdcfae0c1 > 2019-10-13 07:15:09.311406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 15 Tot 1888) > > 35f8e251-1899-41c2-aa77-32b5e5292e3c > 2019-10-13 07:15:55.871412 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [35f8e251-1899-41c2-aa77-32b5e5292e3c] > > 35f8e251-1899-41c2-aa77-32b5e5292e3c > 2019-10-13 07:15:55.871412 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 16 Tot 1889) > > ead16196-c18e-4160-9414-53bc89533d5e > 2019-10-13 07:16:37.511415 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [ead16196-c18e-4160-9414-53bc89533d5e] > > ead16196-c18e-4160-9414-53bc89533d5e > 2019-10-13 07:16:37.511415 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 17 Tot 1890) > > cf3ddbd8-149d-4ce8-b8f1-af2b7eb83c27 > 2019-10-13 07:17:21.991415 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [cf3ddbd8-149d-4ce8-b8f1-af2b7eb83c27] > > cf3ddbd8-149d-4ce8-b8f1-af2b7eb83c27 > 2019-10-13 07:17:21.991415 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 18 Tot 1891) > > 153770e1-e666-4740-bcd9-a27bb78802d3 > 2019-10-13 07:17:57.451412 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [153770e1-e666-4740-bcd9-a27bb78802d3] > > 153770e1-e666-4740-bcd9-a27bb78802d3 > 2019-10-13 07:17:57.451412 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 19 Tot 1892) > >  2019-10-13 07:18:27.271408 [WARNING] > sofia.c:2298 MSG Thread 9 Started > >  2019-10-13 07:18:27.271408 [WARNING] > sofia.c:2298 MSG Thread 8 Started > > f86ad1e8-970d-4412-a0b7-17babd291684 > 2019-10-13 07:18:34.671406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [f86ad1e8-970d-4412-a0b7-17babd291684] > > f86ad1e8-970d-4412-a0b7-17babd291684 > 2019-10-13 07:18:34.671406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 20 Tot 1893) > > f666902f-7d21-4926-99d6-b835c6ccd488 > 2019-10-13 07:19:16.831409 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [f666902f-7d21-4926-99d6-b835c6ccd488] > > f666902f-7d21-4926-99d6-b835c6ccd488 > 2019-10-13 07:19:16.831409 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 21 Tot 1894) > > 02ebf2e0-ffbb-4177-acbc-d5d8ac8b7465 > 2019-10-13 07:19:48.171406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [02ebf2e0-ffbb-4177-acbc-d5d8ac8b7465] > > 02ebf2e0-ffbb-4177-acbc-d5d8ac8b7465 > 2019-10-13 07:19:48.171406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 22 Tot 1895) > > c1969653-a8cb-4c30-88ab-83ec92697576 > 2019-10-13 07:20:36.751414 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [c1969653-a8cb-4c30-88ab-83ec92697576] > > c1969653-a8cb-4c30-88ab-83ec92697576 > 2019-10-13 07:20:36.751414 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 23 Tot 1896) > > 2c7a317e-cf69-4d7b-86e9-3a72ceaf02fe > 2019-10-13 07:21:01.931411 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [2c7a317e-cf69-4d7b-86e9-3a72ceaf02fe] > > 2c7a317e-cf69-4d7b-86e9-3a72ceaf02fe > 2019-10-13 07:21:01.931411 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 24 Tot 1897) > > 51b48636-ca60-417f-800a-376545c78e73 > 2019-10-13 07:21:58.051407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [51b48636-ca60-417f-800a-376545c78e73] > > 51b48636-ca60-417f-800a-376545c78e73 > 2019-10-13 07:21:58.051407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 25 Tot 1898) > > 3622f904-9916-4269-83b1-4b0f3e3e62a2 > 2019-10-13 07:22:17.931407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [3622f904-9916-4269-83b1-4b0f3e3e62a2] > > 3622f904-9916-4269-83b1-4b0f3e3e62a2 > 2019-10-13 07:22:17.931407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 26 Tot 1899) > >  2019-10-13 07:22:18.931414 [WARNING] > sofia.c:2298 MSG Thread 11 Started > >  2019-10-13 07:22:18.931414 [WARNING] > sofia.c:2298 MSG Thread 10 Started > > 9056d310-c7a1-4cd6-80c7-46f87d9a0784 > 2019-10-13 07:23:25.611407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [9056d310-c7a1-4cd6-80c7-46f87d9a0784] > > 9056d310-c7a1-4cd6-80c7-46f87d9a0784 > 2019-10-13 07:23:25.611407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 27 Tot 1900) > > 0a60d031-9dea-4342-8d89-9b803afaaf60 > 2019-10-13 07:23:32.291406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [0a60d031-9dea-4342-8d89-9b803afaaf60] > > 0a60d031-9dea-4342-8d89-9b803afaaf60 > 2019-10-13 07:23:32.291406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 28 Tot 1901) > > a8f29276-38d0-4063-bf03-b9f930665209 > 2019-10-13 07:24:47.171410 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [a8f29276-38d0-4063-bf03-b9f930665209] > > a8f29276-38d0-4063-bf03-b9f930665209 > 2019-10-13 07:24:47.171410 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 29 Tot 1902) > > 5bc8915a-0c46-4623-849b-4394fda212ca > 2019-10-13 07:24:47.571408 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [5bc8915a-0c46-4623-849b-4394fda212ca] > > 5bc8915a-0c46-4623-849b-4394fda212ca > 2019-10-13 07:24:47.571408 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 30 Tot 1903) > > f825e968-9f5c-44ad-bdca-6db6761a2596 > 2019-10-13 07:26:01.751418 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [f825e968-9f5c-44ad-bdca-6db6761a2596] > > f825e968-9f5c-44ad-bdca-6db6761a2596 > 2019-10-13 07:26:01.751418 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 31 Tot 1904) > >  2019-10-13 07:26:04.051411 [WARNING] > sofia.c:2298 MSG Thread 13 Started > >  2019-10-13 07:26:04.051411 [WARNING] > sofia.c:2298 MSG Thread 12 Started > > 979d489d-ff37-413e-b9c9-653b4174bcc6 > 2019-10-13 07:26:07.331412 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [979d489d-ff37-413e-b9c9-653b4174bcc6] > > 979d489d-ff37-413e-b9c9-653b4174bcc6 > 2019-10-13 07:26:07.331412 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 32 Tot 1905) > > b73bab7f-04a8-4f62-914c-2c5efe4db3b5 > 2019-10-13 07:27:17.171406 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [b73bab7f-04a8-4f62-914c-2c5efe4db3b5] > > b73bab7f-04a8-4f62-914c-2c5efe4db3b5 > 2019-10-13 07:27:17.171406 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 33 Tot 1906) > > 62412883-bc04-4095-9a0a-635603642dd6 > 2019-10-13 07:27:28.331417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [62412883-bc04-4095-9a0a-635603642dd6] > > 62412883-bc04-4095-9a0a-635603642dd6 > 2019-10-13 07:27:28.331417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 34 Tot 1907) > > b89eee34-edba-4076-ba07-7c4b1e278793 > 2019-10-13 07:28:33.911403 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [b89eee34-edba-4076-ba07-7c4b1e278793] > > b89eee34-edba-4076-ba07-7c4b1e278793 > 2019-10-13 07:28:33.911403 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 35 Tot 1908) > > 6ae54722-edda-4d4e-aa29-72982e841ab0 > 2019-10-13 07:28:52.171407 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [6ae54722-edda-4d4e-aa29-72982e841ab0] > > 6ae54722-edda-4d4e-aa29-72982e841ab0 > 2019-10-13 07:28:52.171407 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 36 Tot 1909) > >  2019-10-13 07:29:41.031410 [WARNING] > sofia.c:2298 MSG Thread 15 Started > >  2019-10-13 07:29:41.031410 [WARNING] > sofia.c:2298 MSG Thread 14 Started > > a1443915-0e63-45ea-9765-c137d0c7a1d2 > 2019-10-13 07:29:46.811402 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [a1443915-0e63-45ea-9765-c137d0c7a1d2] > > a1443915-0e63-45ea-9765-c137d0c7a1d2 > 2019-10-13 07:29:46.811402 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 37 Tot 1910) > > eee3a285-a32e-4992-924b-e1e2ed8a5dcc > 2019-10-13 07:30:12.411417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [eee3a285-a32e-4992-924b-e1e2ed8a5dcc] > > eee3a285-a32e-4992-924b-e1e2ed8a5dcc > 2019-10-13 07:30:12.411417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 38 Tot 1911) > > 5fc1f582-df14-49a4-bff6-c8d452733367 > 2019-10-13 07:30:59.591415 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/105 at A.B.C.D > [5fc1f582-df14-49a4-bff6-c8d452733367] > > 5fc1f582-df14-49a4-bff6-c8d452733367 > 2019-10-13 07:30:59.591415 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/105 at A.B.C.D) Running > State Change CS_NEW (Cur 39 Tot 1912) > > 8e2a1f5b-bfce-4e98-a04e-416d236a2e86 > 2019-10-13 07:31:33.751417 [NOTICE] > switch_channel.c:1118 New Channel > sofia/sipinterface_2/1001 at A.B.C.D > [8e2a1f5b-bfce-4e98-a04e-416d236a2e86] > > 8e2a1f5b-bfce-4e98-a04e-416d236a2e86 > 2019-10-13 07:31:33.751417 [DEBUG] > switch_core_state_machine.c:585 > (sofia/sipinterface_2/1001 at A.B.C.D) > Running State Change CS_NEW (Cur 40 Tot 1913) > > > >  On Sat, Oct 12, 2019 at 9:13 PM David > Villasmil > > wrote: > > > >  That is a command that doesn’t enables > the logs forever, only for the duration of > the session, once you restart it’s back to > normal. > > > >  But that doesn’t matter. If FS is dying > and restarting or not processing request > l, you should have some logging about it. > > > >  On Sun, 13 Oct 2019 at 00:25, Lloyd > Aloysius > wrote: > > > >  Does the following command enable the > logs for future > >  sofia loglevel all 9 ? > > > >  -- > > > >  On Sat, Oct 12, 2019 at 6:43 PM Seven > Du > wrote: > > > >  run the following command and see if > there's any logs > > > >  sofia loglevel all 9 > > > >  On Sun, Oct 13, 2019 at 6:14 AM Lloyd > Aloysius > wrote: > > > >  Hello > > > >  Currently running the following version. > > > >  FreeSWITCH (Version 1.10.1-release git > f999022 2019-08-20 16:54:04Z 64bit) is ready > > > >  FreeSWITCH Freeze and Fails to process > any traffic. After restart FreeSWITCH.. > Everything ok. > > > >  How to troubleshoot? How to find what > cause this situation. > > > >  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 > > > >  -- > >  About: http://about.me/dujinfang > >  Blog: http://www.dujinfang.com > >  Proj: http://www.freeswitch.org.cn > > > _________________________________________________________________________ > > > >  The FreeSWITCH project is sponsored by > SignalWire https://signalwire.com > >  Enhance your FreeSWITCH install with > disruptive priced SMS and PSTN services. > >  Build your next product on our scalable > cloud platform. > > > >  Join our online community to chat in > real time https://signalwire.community > > > >  Professional FreeSWITCH Services > > sales at freeswitch.com > > > https://freeswitch.com > > > >  Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > >  FreeSWITCH-users mailing list > > FreeSWITCH-users 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 > > > >  -- > >  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 > > > > -- > > Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > > phone: +34669448337 > > [2  ] > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by > SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with > disruptive priced SMS and PSTN services. > > Build your next product on our scalable > cloud platform. > > > > Join our online community to chat in > real time https://signalwire.community > > > > Professional FreeSWITCH Services > > sales at freeswitch.com > > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > -- > Your life is like a penny.  You're going > to lose it.  The question is: > How do > you spend it? > >          John Covici wb2una > covici at ccs.covici.com > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by > SignalWire https://signalwire.com > Enhance your FreeSWITCH install with > disruptive priced SMS and PSTN services. > Build your next product on our scalable > cloud platform. > > Join our online community to chat in real > time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire > https://signalwire.com > Enhance your FreeSWITCH install with disruptive > priced SMS and PSTN services. > Build your next product on our scalable cloud > platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 > > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: From ka.rathod99 at gmail.com Tue Nov 12 12:33:54 2019 From: ka.rathod99 at gmail.com (Abhilash Rathod) Date: Tue, 12 Nov 2019 18:03:54 +0530 Subject: [Freeswitch-users] Audio delay on the conference calls In-Reply-To: <7A3722C6-AE5D-4ED6-88F0-CB8475D2D30D@freeswitch.org> References: <7A3722C6-AE5D-4ED6-88F0-CB8475D2D30D@freeswitch.org> Message-ID: Hello Mike, Thanks for the answer. I am facing this issue in the production environment I have checked the same thing on Freeswitch-Vanilla-1.8.7 but this issue is not reproduced there. So I don't think it's a version-specific issue. Is there any configuration that I need to look into? If you want I can share my Sofia configuration with you. On Tue, Nov 12, 2019 at 1:23 AM Mike Jerris wrote: > Did you try the current release? > > > > On Nov 8, 2019, at 8:10 AM, Abhilash Rathod > wrote: > > > > Hello, > > > > I am using FreeSWITCH version 1.8.7 on Centos 7. I am facing an audio > delay on the conference call. Three users are on the conference call. > > In the initial time, audio is fine, there was no audio delay but audio > delay increasing wisely as call time duration increasing. Below I have > mentioned some delay examples which I have faced. > > > > Call Time: Delay(approx): > > 00:45 1.02s > > 02:30 1.10s > > 04:35 1.25s > > 07:45 1.60s > > 10:35 1.80s > > 15:15 2.60s > > 21:40 3.60s > > 27:10 4.25s > > 30:00 4.30s > > > > I have applied two changes on the FreeSWITCH for the fix this issue, but > still, i am facing audio delay issue. > > 1. set rtp_time_name= none in dialplan > > 2. set in sofia > settings. > > > > Is there any other solution available to resolve this issue. > > Advance Thanks for the help. > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- *Thanks & Regards* *Abhilash Rathod* *[Skype]:- rathod.abhilash* *[Whats App]:- **+91 9016232506* -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Tue Nov 12 19:42:51 2019 From: davidswalkabout at gmail.com (David P) Date: Tue, 12 Nov 2019 11:42:51 -0800 Subject: [Freeswitch-users] Audio stops reaching verto user during call Message-ID: In verto calls since yesterday, I've been able to hear only the first automated prompt played in a conference, even though the mp4 recording reveals later prompts were played. The first prompt I don't hear is at 18:38:58 (based on the clock of another machine in the same AWS cluster), that occurs between these lines from the FS log: 237113d5-a985-4527-9d2c-67a3681b5202 2019-11-12 18:37:27.385974 [DEBUG] switch_rtp.c:8187 Correct audio ip/port confirmed. 2019-11-12 18:40:00.685949 [ALERT] mod_verto.c:1411 READ ((public IP of FS)):60168 [{ "jsonrpc": "2.0", "method": "verto.bye", Our server has been handling calls well for months. It's FreeSWITCH Version 20.19.4-release-12-fc9d51c~64bit (-release-12-fc9d51c 64bit) on debian9. The call used a relay candidate from our coturn server, and its log shows nothing obviously wrong during this approximate period, but due to its lack of timestamps, it seems practically impossible to align its session info with FS's. I haven't found a way to relate ice IP numbers in the FS log with IP numbers in coturn's. Before I turn to the coturn mailing list for help, is there anything on the FS side you can suggest? -------------- next part -------------- An HTML attachment was scrubbed... URL: From dgreenwald at gmail.com Wed Nov 13 18:03:45 2019 From: dgreenwald at gmail.com (Daniel Greenwald) Date: Wed, 13 Nov 2019 13:03:45 -0500 Subject: [Freeswitch-users] Sending t38 No-Signal packets while receiving a fax page with mod_spandsp In-Reply-To: References: Message-ID: Anyone know if freeswitch can be made to send periodic No-signal packets while receiving a page data pump? Carrier's SBC is hitting an RTP timer during long page transfers because freeswitch/spandsp is completely silent while receiving a page. Other T38 switches I've noticed will send a No-Signal packet every 300ms seconds as a keep-alive. This can be seen in the top screenshot below. Lower one is freeswitch receiving a page silently. [image: image.png] [image: image.png] -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 189606 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 221640 bytes Desc: not available URL: From info at behrend-cs.de Tue Nov 12 21:36:46 2019 From: info at behrend-cs.de (Walter Behrend) Date: Tue, 12 Nov 2019 22:36:46 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster Message-ID: <031c01d599a1$48ec5590$dac500b0$@behrend-cs.de> Hi Seb + list, I think I solved my problem. Thanks to your hint regarding openssl.cnf. I don't know what happened, but as soon as I changed the MinProtocol parameter there to TLSv1 and restarted freeswitch, everything went smooth. Unfortunately, I do not remember if the value was TLSv1.3 or TLSv1.2 previously, I just can tell you it never accepted anything below TLS 1.3 Looks now this way: [system_default_sect] MinProtocol = TLSv1 CipherString = DEFAULT at SECLEVEL=2 Everything else is like shown in your link to the diff. Now, everything >= TLS1.0 is accepted Thank you so much! Cheers, Walter On Tue, Nov 12, 2019 at 09:27:16PM +0100, Sebastian Kemper wrote: > On Mon, Nov 11, 2019 at 11:21:05PM +0100, Walter Behrend wrote: > > It seems it does not matter which value I set for "tls_version" - in > > every case, my TLS enabled port only accepts TLS 1.3 connections. I > > have the problem that we're also using older phones which only support TLS 1.0. > > > > Error message is: > > > > tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL > > routines:tls_early_post_process_client_hello:unsupported protocol > > Works fine here: > > 4 0.004770 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello > 6 0.193706 192.168.0.1 → 192.168.0.120 TLSv1.2 1514 Server Hello > 7 0.193809 192.168.0.1 → 192.168.0.120 TLSv1.2 871 Certificate, Server Key Exchange, Server Hello Done > 10 0.256056 192.168.0.120 → 192.168.0.1 TLSv1.2 141 Client Key Exchange > 12 0.269076 192.168.0.120 → 192.168.0.1 TLSv1.2 72 Change Cipher Spec > 14 0.269344 192.168.0.120 → 192.168.0.1 TLSv1.2 103 Encrypted Handshake Message > > With openssl s_client I can also connect with TLS1.0, 1.1, 1.2 and 1.3 > (which suggests that FS isn't using system openssl config). Actually it doesn't work fine here after all. I had updated /etc/ssl/openssl.cnf with the Debian changes, but actually openssl_conf = default_conf was overwritten by some OpenWrt config snippet later. I amended that and now when I set tls_version to something specific I get 4 0.010062 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010927 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Protocol Version) and when I leave it unset I get 4 0.009440 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010884 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Handshake Failure) Regards, Seb From info at behrend-cs.de Tue Nov 12 21:38:40 2019 From: info at behrend-cs.de (Walter Behrend) Date: Tue, 12 Nov 2019 22:38:40 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster In-Reply-To: <20191112211454.GA3609@darth.lan> References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> <20191112202715.GA3584@darth.lan> <20191112211454.GA3609@darth.lan> Message-ID: <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de> Btw, I think there is a problem in freeswitch - if for example I configure stunnel, there is no problem with specifying accepting also older TLS standards without the need of changing the MinProtocol setting within the openssl.cnf file. As a user or admin, I would normally expect the tls-version parameter to do the same job for me... Cheers, Walter On Tue, Nov 12, 2019 at 09:27:16PM +0100, Sebastian Kemper wrote: > On Mon, Nov 11, 2019 at 11:21:05PM +0100, Walter Behrend wrote: > > It seems it does not matter which value I set for "tls_version" - in > > every case, my TLS enabled port only accepts TLS 1.3 connections. I > > have the problem that we're also using older phones which only support TLS 1.0. > > > > Error message is: > > > > tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL > > routines:tls_early_post_process_client_hello:unsupported protocol > > Works fine here: > > 4 0.004770 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello > 6 0.193706 192.168.0.1 → 192.168.0.120 TLSv1.2 1514 Server Hello > 7 0.193809 192.168.0.1 → 192.168.0.120 TLSv1.2 871 Certificate, Server Key Exchange, Server Hello Done > 10 0.256056 192.168.0.120 → 192.168.0.1 TLSv1.2 141 Client Key Exchange > 12 0.269076 192.168.0.120 → 192.168.0.1 TLSv1.2 72 Change Cipher Spec > 14 0.269344 192.168.0.120 → 192.168.0.1 TLSv1.2 103 Encrypted Handshake Message > > With openssl s_client I can also connect with TLS1.0, 1.1, 1.2 and 1.3 > (which suggests that FS isn't using system openssl config). Actually it doesn't work fine here after all. I had updated /etc/ssl/openssl.cnf with the Debian changes, but actually openssl_conf = default_conf was overwritten by some OpenWrt config snippet later. I amended that and now when I set tls_version to something specific I get 4 0.010062 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010927 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Protocol Version) and when I leave it unset I get 4 0.009440 192.168.0.120 → 192.168.0.1 TLSv1 464 Client Hello 6 0.010884 192.168.0.1 → 192.168.0.120 TLSv1.2 73 Alert (Level: Fatal, Description: Handshake Failure) Regards, Seb From mike at freeswitch.org Wed Nov 13 19:25:05 2019 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 13 Nov 2019 11:25:05 -0800 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: References: Message-ID: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> What rate is your conference running at? > On Nov 12, 2019, at 6:49 PM, 王聡 wrote: > > Hey all, > > I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: > > session:execute("pre_answer") > session:execute("conference_set_auto_outcall", "user/" .. args.call_user) > session:execute("conference", "testroom at default") > > My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: > > variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 > variable_rtp_use_codec_name: opus > variable_rtp_use_codec_fmtp: useinbandfec%3D1 > variable_rtp_use_codec_rate: 48000 > variable_rtp_use_codec_ptime: 20 > variable_rtp_use_codec_channels: 1 > variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c > variable_original_read_codec: opus > variable_write_codec: opus > variable_read_codec: opus > > But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: > > variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 > variable_rtp_use_codec_name: opus > variable_rtp_use_codec_fmtp: useinbandfec%3D1 > variable_rtp_use_codec_rate: 48000 > variable_rtp_use_codec_ptime: 20 > variable_rtp_use_codec_channels: 1 > variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c > variable_original_read_codec: opus > variable_write_codec: opus > variable_read_codec: L16 > > Both tests are based on Linphone offical app. > During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. > Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. > Any suggestion would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Wed Nov 13 19:27:10 2019 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 13 Nov 2019 11:27:10 -0800 Subject: [Freeswitch-users] Audio delay on the conference calls In-Reply-To: References: <7A3722C6-AE5D-4ED6-88F0-CB8475D2D30D@freeswitch.org> Message-ID: Does the same thing happen on the current 1.10 release? > On Nov 12, 2019, at 4:33 AM, Abhilash Rathod wrote: > > Hello Mike, > > Thanks for the answer. > I am facing this issue in the production environment I have checked the same thing on Freeswitch-Vanilla-1.8.7 but this issue is not reproduced there. > So I don't think it's a version-specific issue. > > Is there any configuration that I need to look into? > If you want I can share my Sofia configuration with you. > > > On Tue, Nov 12, 2019 at 1:23 AM Mike Jerris > wrote: > Did you try the current release? > > > > On Nov 8, 2019, at 8:10 AM, Abhilash Rathod > wrote: > > > > Hello, > > > > I am using FreeSWITCH version 1.8.7 on Centos 7. I am facing an audio delay on the conference call. Three users are on the conference call. > > In the initial time, audio is fine, there was no audio delay but audio delay increasing wisely as call time duration increasing. Below I have mentioned some delay examples which I have faced. > > > > Call Time: Delay(approx): > > 00:45 1.02s > > 02:30 1.10s > > 04:35 1.25s > > 07:45 1.60s > > 10:35 1.80s > > 15:15 2.60s > > 21:40 3.60s > > 27:10 4.25s > > 30:00 4.30s > > > > I have applied two changes on the FreeSWITCH for the fix this issue, but still, i am facing audio delay issue. > > 1. set rtp_time_name= none in dialplan > > 2. set in sofia settings. > > > > Is there any other solution available to resolve this issue. > > Advance Thanks for the help. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From coppice12 at gmail.com Wed Nov 13 19:44:57 2019 From: coppice12 at gmail.com (Steve Underwood) Date: Wed, 13 Nov 2019 19:44:57 +0000 Subject: [Freeswitch-users] Sending t38 No-Signal packets while receiving a fax page with mod_spandsp In-Reply-To: References: Message-ID: On 11/13/19 6:03 PM, Daniel Greenwald wrote: > Anyone know if freeswitch can be made to send periodic No-signal > packets while receiving a page data pump? Carrier's SBC is hitting an > RTP timer during long page transfers because freeswitch/spandsp is > completely silent while receiving a page. Other T38 switches I've > noticed will send a No-Signal packet every 300ms seconds as a > keep-alive. This can be seen in the top screenshot below. Lower one is > freeswitch receiving a page silently. So, you've found yet another broken buggy system that doesn't follow standards. Surprise, surprise. If you find the person who made the system require no signal packets every 300ms can you find a 4x2 and wack them around the head with it for me, please? There is an option in spandsp itself to make it regularly repeat the indicator signals, to tolerate this kind of broken behaviour. I'm not sure if the Freeswitch module exposes that capability. Regards, Steve From brian at freeswitch.com Wed Nov 13 19:55:47 2019 From: brian at freeswitch.com (Brian West) Date: Wed, 13 Nov 2019 11:55:47 -0800 Subject: [Freeswitch-users] Sending t38 No-Signal packets while receiving a fax page with mod_spandsp In-Reply-To: References: Message-ID: Send me the details and I can expose that. On Wed, Nov 13, 2019 at 11:45 AM Steve Underwood wrote: > On 11/13/19 6:03 PM, Daniel Greenwald wrote: > > Anyone know if freeswitch can be made to send periodic No-signal > > packets while receiving a page data pump? Carrier's SBC is hitting an > > RTP timer during long page transfers because freeswitch/spandsp is > > completely silent while receiving a page. Other T38 switches I've > > noticed will send a No-Signal packet every 300ms seconds as a > > keep-alive. This can be seen in the top screenshot below. Lower one is > > freeswitch receiving a page silently. > > So, you've found yet another broken buggy system that doesn't follow > standards. Surprise, surprise. If you find the person who made the > system require no signal packets every 300ms can you find a 4x2 and wack > them around the head with it for me, please? > > There is an option in spandsp itself to make it regularly repeat the > indicator signals, to tolerate this kind of broken behaviour. I'm not > sure if the Freeswitch module exposes that capability. > > Regards, > > Steve > > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From sebastian_ml at gmx.net Wed Nov 13 20:01:31 2019 From: sebastian_ml at gmx.net (Sebastian Kemper) Date: Wed, 13 Nov 2019 21:01:31 +0100 Subject: [Freeswitch-users] Problems with TLS after upgrading to Buster In-Reply-To: <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de> References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> <20191112202715.GA3584@darth.lan> <20191112211454.GA3609@darth.lan> <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de> Message-ID: <20191113200131.GA738@darth.lan> On Tue, Nov 12, 2019 at 10:38:40PM +0100, Walter Behrend wrote: > Btw, I think there is a problem in freeswitch - if for example I > configure stunnel, there is no problem with specifying accepting also > older TLS standards without the need of changing the MinProtocol > setting within the openssl.cnf file. As a user or admin, I would > normally expect the tls-version parameter to do the same job for me... Hi Walter, I guess that's a point of view. I was quite happy to find that OpenSSL enforces the restrictions set in /etc/ssl/openssl.cnf also when used through FreeSWITCH. I'd find it rather strange if it didn't, honestly. If they're the default settings then they have to be enforced whenever OpenSSL is used, in my opinion. I've tested with an updated message digest in gentls_cert (SHA256 like you suggested) and can confirm it's working properly with this. I've sent a pull request via GitHub to FS. Regards, Seb From michael at mailworks.org Thu Nov 14 04:31:49 2019 From: michael at mailworks.org (Michael Avers) Date: Wed, 13 Nov 2019 21:31:49 -0700 Subject: [Freeswitch-users] presence_id in dial-string is not behaving correctly In-Reply-To: <20191113200131.GA738@darth.lan> References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> <20191112202715.GA3584@darth.lan> <20191112211454.GA3609@darth.lan> <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de> <20191113200131.GA738@darth.lan> Message-ID: Hello, We have a need to set presence_id to an extension different than the one being dialed. We do this in the dial-string of the dialed extension: Say dialed_user is 3005, we set presence_id to be for 3001 with this dial-string: {presence_id=3001@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})} When 3005 is dialed, 3001's presence indicator on the phones (in this case Yealink) does indeed behave correctly, that is until an answered call is hung up - in that case the presence is not cleared and the BLF subscription for 3001 at domain remains lit up. Anyone seen this behavior before? Is there a better way to set a custom presence_id for extensions? This was tested on FS 1.6.20 as well as latest master just now. Thank you, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Thu Nov 14 05:14:34 2019 From: michael at mailworks.org (Michael Avers) Date: Wed, 13 Nov 2019 22:14:34 -0700 Subject: [Freeswitch-users] =?utf-8?q?presence=5Fid_in_dial-string_is_not_?= =?utf-8?q?behaving=09correctly?= In-Reply-To: References: <020001d598de$4fb28980$ef179c80$@behrend-cs.de> <20191112202715.GA3584@darth.lan> <20191112211454.GA3609@darth.lan> <031d01d599a1$8d30aee0$a7920ca0$@behrend-cs.de> <20191113200131.GA738@darth.lan> Message-ID: <18d9e412-e213-489b-8cfd-882703129e92@www.fastmail.com> Answering my own question -- oops, I overlooked the presence_id directory variable. Setting that to 3001 at domain for the 3005 user directory resolved the issue. Mike On Wed, Nov 13, 2019, at 9:31 PM, Michael Avers wrote: > Hello, > > We have a need to set presence_id to an extension different than the one being dialed. We do this in the dial-string of the dialed extension: > > Say dialed_user is 3005, we set presence_id to be for 3001 with this dial-string: > > {presence_id=3001@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})} > > When 3005 is dialed, 3001's presence indicator on the phones (in this case Yealink) does indeed behave correctly, that is until an answered call is hung up - in that case the presence is not cleared and the BLF subscription for 3001 at domain remains lit up. > > Anyone seen this behavior before? Is there a better way to set a custom presence_id for extensions? > > This was tested on FS 1.6.20 as well as latest master just now. > > Thank you, > Mike > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 cong.wang.itsherpa at gmail.com Thu Nov 14 09:42:23 2019 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Thu, 14 Nov 2019 18:42:23 +0900 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> References: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> Message-ID: Perhaps 48000, from uuid_dump: variable_rtp_use_codec_rate: 48000 Regards. > 在 2019年11月14日,04:25,Mike Jerris 写道: > > What rate is your conference running at? > >> On Nov 12, 2019, at 6:49 PM, 王聡 > wrote: >> >> Hey all, >> >> I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: >> >> session:execute("pre_answer") >> session:execute("conference_set_auto_outcall", "user/" .. args.call_user) >> session:execute("conference", "testroom at default") >> >> My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: >> >> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >> variable_rtp_use_codec_name: opus >> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >> variable_rtp_use_codec_rate: 48000 >> variable_rtp_use_codec_ptime: 20 >> variable_rtp_use_codec_channels: 1 >> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >> variable_original_read_codec: opus >> variable_write_codec: opus >> variable_read_codec: opus >> >> But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: >> >> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >> variable_rtp_use_codec_name: opus >> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >> variable_rtp_use_codec_rate: 48000 >> variable_rtp_use_codec_ptime: 20 >> variable_rtp_use_codec_channels: 1 >> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >> variable_original_read_codec: opus >> variable_write_codec: opus >> variable_read_codec: L16 >> >> Both tests are based on Linphone offical app. >> During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. >> Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. >> Any suggestion would be appreciated. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 dgreenwald at gmail.com Thu Nov 14 13:56:20 2019 From: dgreenwald at gmail.com (Daniel Greenwald) Date: Thu, 14 Nov 2019 08:56:20 -0500 Subject: [Freeswitch-users] Sending t38 No-Signal packets while receiving a fax page with mod_spandsp In-Reply-To: References: Message-ID: Wow Steve that's amazing! Can you point us to the option so we can see about enabling it in freeswitch? I've been looking through the spandsp source and unsure where this flag is. We just need to send some packets to prevent the 5 minute one-way RTP timer kicking in. Thanks! On Wed, Nov 13, 2019 at 3:31 PM Brian West wrote: > Send me the details and I can expose that. > > On Wed, Nov 13, 2019 at 11:45 AM Steve Underwood > wrote: > >> On 11/13/19 6:03 PM, Daniel Greenwald wrote: >> > Anyone know if freeswitch can be made to send periodic No-signal >> > packets while receiving a page data pump? Carrier's SBC is hitting an >> > RTP timer during long page transfers because freeswitch/spandsp is >> > completely silent while receiving a page. Other T38 switches I've >> > noticed will send a No-Signal packet every 300ms seconds as a >> > keep-alive. This can be seen in the top screenshot below. Lower one is >> > freeswitch receiving a page silently. >> >> So, you've found yet another broken buggy system that doesn't follow >> standards. Surprise, surprise. If you find the person who made the >> system require no signal packets every 300ms can you find a 4x2 and wack >> them around the head with it for me, please? >> >> There is an option in spandsp itself to make it regularly repeat the >> indicator signals, to tolerate this kind of broken behaviour. I'm not >> sure if the Freeswitch module exposes that capability. >> >> Regards, >> >> Steve >> >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > > Brian West | Co-founder and Developer > > Need Commercial support? email sales at freeswitch.com > > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > > > Email: brian at freeswitch.com > > Mobile: 918-424-9378 > > Website: https://www.FreeSWITCH.com > > [image: https://www.facebook.com/signalwireinc?src=email] > [image: > https://twitter.com/freeswitch] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Nov 14 15:54:06 2019 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 14 Nov 2019 15:54:06 +0000 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: References: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> Message-ID: Your issue does not have anything to do with L16 per se, and you cannot disable it for mod_conference. Everything is decoded into L16 / encoded from L16, like all audio codecs do. A conference is more CPU intensive than a bridge. On Thu, Nov 14, 2019 at 9:43 AM 王聡 wrote: > Perhaps 48000, from uuid_dump: > > variable_rtp_use_codec_rate: 48000 > > > Regards. > > 在 2019年11月14日,04:25,Mike Jerris 写道: > > What rate is your conference running at? > > On Nov 12, 2019, at 6:49 PM, 王聡 wrote: > > Hey all, > > I’m trying to transfer my ivr system from bridge to conference for more > function. The simple conference ivr likes: > > session:execute("pre_answer") > session:execute("conference_set_auto_outcall", "user/" .. args.call_user) > session:execute("conference", "testroom at default") > > > My FreeSWITCH server is configured to accept only opus for audio codec, > and it worked well on bridge mode. The uuid_dump showed: > > variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 > variable_rtp_use_codec_name: opus > variable_rtp_use_codec_fmtp: useinbandfec%3D1 > variable_rtp_use_codec_rate: 48000 > variable_rtp_use_codec_ptime: 20 > variable_rtp_use_codec_channels: 1 > variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c > variable_original_read_codec: opus > variable_write_codec: opus > variable_read_codec: opus > > > But after I modified my ivr into conference, the read_codec turned into > L16. Uuid_dump showed: > > variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 > variable_rtp_use_codec_name: opus > variable_rtp_use_codec_fmtp: useinbandfec%3D1 > variable_rtp_use_codec_rate: 48000 > variable_rtp_use_codec_ptime: 20 > variable_rtp_use_codec_channels: 1 > variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c > variable_original_read_codec: opus > variable_write_codec: opus > variable_read_codec: L16 > > > Both tests are based on Linphone offical app. > During test, the audio quality has a significant loss on conference mode > compared with bridge, perhaps due to L16 codec. > Is there any solutions for this situation? I wonder if I can disable L16 > on my server, or force the codec to opus in conference. > Any suggestion would be appreciated. > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 mike at freeswitch.org Thu Nov 14 18:26:29 2019 From: mike at freeswitch.org (Mike Jerris) Date: Thu, 14 Nov 2019 10:26:29 -0800 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: References: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> Message-ID: No, thats the codec rate. This will be in your conference configuration. > On Nov 14, 2019, at 1:42 AM, 王聡 wrote: > > Perhaps 48000, from uuid_dump: > > variable_rtp_use_codec_rate: 48000 > > Regards. > >> 在 2019年11月14日,04:25,Mike Jerris > 写道: >> >> What rate is your conference running at? >> >>> On Nov 12, 2019, at 6:49 PM, 王聡 > wrote: >>> >>> Hey all, >>> >>> I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: >>> >>> session:execute("pre_answer") >>> session:execute("conference_set_auto_outcall", "user/" .. args.call_user) >>> session:execute("conference", "testroom at default") >>> >>> My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: >>> >>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>> variable_rtp_use_codec_name: opus >>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>> variable_rtp_use_codec_rate: 48000 >>> variable_rtp_use_codec_ptime: 20 >>> variable_rtp_use_codec_channels: 1 >>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>> variable_original_read_codec: opus >>> variable_write_codec: opus >>> variable_read_codec: opus >>> >>> But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: >>> >>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>> variable_rtp_use_codec_name: opus >>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>> variable_rtp_use_codec_rate: 48000 >>> variable_rtp_use_codec_ptime: 20 >>> variable_rtp_use_codec_channels: 1 >>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>> variable_original_read_codec: opus >>> variable_write_codec: opus >>> variable_read_codec: L16 >>> >>> Both tests are based on Linphone offical app. >>> During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. >>> Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. >>> Any suggestion would be appreciated. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From cong.wang.itsherpa at gmail.com Fri Nov 15 04:10:29 2019 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Fri, 15 Nov 2019 13:10:29 +0900 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: References: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> Message-ID: <4D1D748B-0D14-4B47-A615-1093D9E18FC6@gmail.com> Thanks for your reply. However we had found the audio quality under conference mod is not as good as bridge mode, though both OPUS. I wonder if this issue has any relations on L16 or not. Regards. > 在 2019年11月15日,00:54,Dragos Oancea 写道: > > Your issue does not have anything to do with L16 per se, and you cannot disable it for mod_conference. Everything is decoded into L16 / encoded from L16, like all audio codecs do. > A conference is more CPU intensive than a bridge. > > On Thu, Nov 14, 2019 at 9:43 AM 王聡 > wrote: > Perhaps 48000, from uuid_dump: > > variable_rtp_use_codec_rate: 48000 > > Regards. > >> 在 2019年11月14日,04:25,Mike Jerris > 写道: >> >> What rate is your conference running at? >> >>> On Nov 12, 2019, at 6:49 PM, 王聡 > wrote: >>> >>> Hey all, >>> >>> I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: >>> >>> session:execute("pre_answer") >>> session:execute("conference_set_auto_outcall", "user/" .. args.call_user) >>> session:execute("conference", "testroom at default") >>> >>> My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: >>> >>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>> variable_rtp_use_codec_name: opus >>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>> variable_rtp_use_codec_rate: 48000 >>> variable_rtp_use_codec_ptime: 20 >>> variable_rtp_use_codec_channels: 1 >>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>> variable_original_read_codec: opus >>> variable_write_codec: opus >>> variable_read_codec: opus >>> >>> But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: >>> >>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>> variable_rtp_use_codec_name: opus >>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>> variable_rtp_use_codec_rate: 48000 >>> variable_rtp_use_codec_ptime: 20 >>> variable_rtp_use_codec_channels: 1 >>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>> variable_original_read_codec: opus >>> variable_write_codec: opus >>> variable_read_codec: L16 >>> >>> Both tests are based on Linphone offical app. >>> During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. >>> Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. >>> Any suggestion would be appreciated. >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users 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 cong.wang.itsherpa at gmail.com Fri Nov 15 04:11:48 2019 From: cong.wang.itsherpa at gmail.com (=?utf-8?B?546L6IGh?=) Date: Fri, 15 Nov 2019 13:11:48 +0900 Subject: [Freeswitch-users] L16 Codec in mod_conference In-Reply-To: References: <0F02EFE1-3EFF-4DA5-B275-1BE075CCA68A@freeswitch.org> Message-ID: <8422719B-E5AA-45C4-B3E4-960AEEF79D8D@gmail.com> We hadn’t configure this part, perhaps default? Sorry I’m newcomer to mod_conference. Regards. > 在 2019年11月15日,03:26,Mike Jerris 写道: > > No, thats the codec rate. This will be in your conference configuration. > >> On Nov 14, 2019, at 1:42 AM, 王聡 > wrote: >> >> Perhaps 48000, from uuid_dump: >> >> variable_rtp_use_codec_rate: 48000 >> >> Regards. >> >>> 在 2019年11月14日,04:25,Mike Jerris > 写道: >>> >>> What rate is your conference running at? >>> >>>> On Nov 12, 2019, at 6:49 PM, 王聡 > wrote: >>>> >>>> Hey all, >>>> >>>> I’m trying to transfer my ivr system from bridge to conference for more function. The simple conference ivr likes: >>>> >>>> session:execute("pre_answer") >>>> session:execute("conference_set_auto_outcall", "user/" .. args.call_user) >>>> session:execute("conference", "testroom at default") >>>> >>>> My FreeSWITCH server is configured to accept only opus for audio codec, and it worked well on bridge mode. The uuid_dump showed: >>>> >>>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>>> variable_rtp_use_codec_name: opus >>>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>>> variable_rtp_use_codec_rate: 48000 >>>> variable_rtp_use_codec_ptime: 20 >>>> variable_rtp_use_codec_channels: 1 >>>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>>> variable_original_read_codec: opus >>>> variable_write_codec: opus >>>> variable_read_codec: opus >>>> >>>> But after I modified my ivr into conference, the read_codec turned into L16. Uuid_dump showed: >>>> >>>> variable_rtp_use_codec_string: OPUS,H264,H263-1998,VP8 >>>> variable_rtp_use_codec_name: opus >>>> variable_rtp_use_codec_fmtp: useinbandfec%3D1 >>>> variable_rtp_use_codec_rate: 48000 >>>> variable_rtp_use_codec_ptime: 20 >>>> variable_rtp_use_codec_channels: 1 >>>> variable_rtp_last_audio_codec_string: opus%4048000h%4020i%401c >>>> variable_original_read_codec: opus >>>> variable_write_codec: opus >>>> variable_read_codec: L16 >>>> >>>> Both tests are based on Linphone offical app. >>>> During test, the audio quality has a significant loss on conference mode compared with bridge, perhaps due to L16 codec. >>>> Is there any solutions for this situation? I wonder if I can disable L16 on my server, or force the codec to opus in conference. >>>> Any suggestion would be appreciated. >>> > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From tanim at surroundapps.com Fri Nov 15 15:31:03 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Fri, 15 Nov 2019 21:31:03 +0600 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection Message-ID: Hi I have originated outbound call to pstn number using gateway provider flowroute with originate command. I am using python ESL to achieve this. If the call is not answered by pstn user, then i need to post status to my webserver . Can anyone suggests me how i can detect whether the pstn call is answered or not? Thanks Tanim -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Fri Nov 15 16:24:55 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Fri, 15 Nov 2019 11:24:55 -0500 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: Tanim -- Since you are using ESL, you should monitor the status of the outbound channel you've originated -- you should do that by channel ID, either the generated one or you could preset it via the origination_uuid variable. During the channel's execution you will see a bunch of CHANNEL_* events streamed via ESL; the relevant event for your question is CHANNEL_HANGUP. You can extract the reason behind the hangup from the Hangup-Cause header (could be CALL_REJECTED, USER_BUSY etc. here's a comprehensive list https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table ). Conversely, if the call is picked up, you will see a CHANNEL_ANSWER event coming through. On Fri, 15 Nov 2019 at 10:59, Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi > > I have originated outbound call to pstn number using gateway provider > flowroute with originate command. I am using python ESL to achieve this. > > If the call is not answered by pstn user, then i need to post status to my > webserver . > > Can anyone suggests me how i can detect whether the pstn call is answered > or not? > > Thanks > Tanim > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Nov 15 16:39:00 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 15 Nov 2019 16:39:00 +0000 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: You’re already using ESL to originate. You can stay connected and filter events to get the event for your call. On Fri, 15 Nov 2019 at 15:53, Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi > > I have originated outbound call to pstn number using gateway provider > flowroute with originate command. I am using python ESL to achieve this. > > If the call is not answered by pstn user, then i need to post status to my > webserver . > > Can anyone suggests me how i can detect whether the pstn call is answered > or not? > > Thanks > Tanim > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From tanim at surroundapps.com Fri Nov 15 18:42:38 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Sat, 16 Nov 2019 00:42:38 +0600 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: Hi Ciprian Thanks for your prompt response. As i am new to freeswitch,can you give some guideline/sample about how i can monitor status of the originated outbound channel? Thanks Tanim On Fri, Nov 15, 2019, 10:24 PM Ciprian Dosoftei wrote: > Tanim -- > > Since you are using ESL, you should monitor the status of the outbound > channel you've originated -- you should do that by channel ID, either the > generated one or you could preset it via the origination_uuid variable. > > During the channel's execution you will see a bunch of CHANNEL_* events > streamed via ESL; the relevant event for your question is CHANNEL_HANGUP. > You can extract the reason behind the hangup from the Hangup-Cause header > (could be CALL_REJECTED, USER_BUSY etc. here's a comprehensive list > https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table > ). > > Conversely, if the call is picked up, you will see a CHANNEL_ANSWER event > coming through. > > > On Fri, 15 Nov 2019 at 10:59, Md,Mehedi Hasan Kabir(Tanim) < > tanim at surroundapps.com> wrote: > >> Hi >> >> I have originated outbound call to pstn number using gateway provider >> flowroute with originate command. I am using python ESL to achieve this. >> >> If the call is not answered by pstn user, then i need to post status to >> my webserver . >> >> Can anyone suggests me how i can detect whether the pstn call is answered >> or not? >> >> Thanks >> Tanim >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From tanim at surroundapps.com Fri Nov 15 18:46:06 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Sat, 16 Nov 2019 00:46:06 +0600 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: Hi David Thanks for your reply. As i am new to freeswitch, can you give some guidelines/sample about how i can stay connected and filter events to get the events for the call? Thanks Tanim On Fri, Nov 15, 2019, 10:39 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > You’re already using ESL to originate. You can stay connected and filter > events to get the event for your call. > > > On Fri, 15 Nov 2019 at 15:53, Md,Mehedi Hasan Kabir(Tanim) < > tanim at surroundapps.com> wrote: > >> Hi >> >> I have originated outbound call to pstn number using gateway provider >> flowroute with originate command. I am using python ESL to achieve this. >> >> If the call is not answered by pstn user, then i need to post status to >> my webserver . >> >> Can anyone suggests me how i can detect whether the pstn call is answered >> or not? >> >> Thanks >> Tanim >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -- > 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 davidswalkabout at gmail.com Fri Nov 15 19:11:54 2019 From: davidswalkabout at gmail.com (David P) Date: Fri, 15 Nov 2019 11:11:54 -0800 Subject: [Freeswitch-users] Building clients for different OSs Message-ID: There was an interesting session at KrankyGeek just now where it was mentioned that Pion could be used to create a WebRTC client for mobile OSs and for browsers via WebAssembly. Is there a verto-like sdk for "write-once" client apps that can then be built for different platforms? -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Nov 15 19:12:13 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 15 Nov 2019 19:12:13 +0000 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: This should go a long way https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=1048914#content/view/1048916 On Fri, 15 Nov 2019 at 19:01, Md,Mehedi Hasan Kabir(Tanim) < tanim at surroundapps.com> wrote: > Hi Ciprian > > Thanks for your prompt response. As i am new to freeswitch,can you give > some guideline/sample about how i can monitor status of the originated > outbound channel? > > Thanks > Tanim > > On Fri, Nov 15, 2019, 10:24 PM Ciprian Dosoftei < > ciprian.dosoftei at gmail.com> wrote: > >> Tanim -- >> >> Since you are using ESL, you should monitor the status of the outbound >> channel you've originated -- you should do that by channel ID, either the >> generated one or you could preset it via the origination_uuid variable. >> >> During the channel's execution you will see a bunch of CHANNEL_* events >> streamed via ESL; the relevant event for your question is CHANNEL_HANGUP. >> You can extract the reason behind the hangup from the Hangup-Cause header >> (could be CALL_REJECTED, USER_BUSY etc. here's a comprehensive list >> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table >> ). >> >> Conversely, if the call is picked up, you will see a CHANNEL_ANSWER event >> coming through. >> >> >> On Fri, 15 Nov 2019 at 10:59, Md,Mehedi Hasan Kabir(Tanim) < >> tanim at surroundapps.com> wrote: >> >>> Hi >>> >>> I have originated outbound call to pstn number using gateway provider >>> flowroute with originate command. I am using python ESL to achieve this. >>> >>> If the call is not answered by pstn user, then i need to post status to >>> my webserver . >>> >>> Can anyone suggests me how i can detect whether the pstn call is >>> answered or not? >>> >>> Thanks >>> Tanim >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Best Regards, >> Ciprian Dosoftei >> >> The information transmitted is intended only for the addressee and may >> contain privileged and/or confidential material. If you are not the >> intended recipient, kindly contact the sender and delete the message. >> >> Any disclosure, distribution or copying of this message is strictly >> prohibited without the expressed permission of the sender. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 hyavari at rocketmail.com Sat Nov 16 12:34:55 2019 From: hyavari at rocketmail.com (H Yavari) Date: Sat, 16 Nov 2019 12:34:55 +0000 (UTC) Subject: Using Curl in dial plan References: <1146766403.1876813.1573907695705.ref@mail.yahoo.com> Message-ID: <1146766403.1876813.1573907695705@mail.yahoo.com> Hi to all, I am going to use curl and post some info like CID after call hang up to another web server. Using the Lua and working with events is the only solution or we have access to call variables in the dial plan when call is finished or voicemail is done? Some useful hints would be appreciated. Regards.H.Yavari -------------- next part -------------- An HTML attachment was scrubbed... URL: From josefu at gmail.com Sat Nov 16 12:37:30 2019 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Sat, 16 Nov 2019 13:37:30 +0100 Subject: [Freeswitch-users] Performance questions Message-ID: Hi, I have several nodes with FreeSWICH, typically 2 socket Xeon v2 (with 8-12 cores every socket) and 64~128GB of RAM. I'm testing different configurations for max performance. I run FreeSWITCH containers (with docker and macvlan network), the base system is Ubuntu 18.04 and I run 4 containers for machine (max concurrency about 500-700 calls and max cps 200, limited in FreeSWITCH config). But I have some questions about the base config of my servers: 1. Is better cpu hyperthreading enabled or disabled? 2. Is relevant NUMA placement for FreeSWITCH (now I bind 2 FreeSWITCH processes to one NUMA node of the machine)? 3. Is it recommended cpu pinning to avoid excessive context switching? (my opinion is yes, but I don't know if the performance difference is too high) 4. I run default Ubuntu kernel (generic ubuntu flavor, with kernel CONFIG_HZ_250=y), but I can change to the "low latency" version, with CONFIG_HZ=1000. Witch is better? I have tested the two versions with timer_test command in FreeSWITCH and I haven't seen any differences (with no load): Avg: 20.000ms Total Time: 1000.002ms Any other advise about network tunning? PD: the containers are Debian Buster with FreeSWITCH 1.10.1 Best regards -- Jose From imfanee at gmail.com Sat Nov 16 13:17:19 2019 From: imfanee at gmail.com (Faisal Hanif) Date: Sat, 16 Nov 2019 18:17:19 +0500 Subject: [Freeswitch-users] Using Curl in dial plan In-Reply-To: References: <1146766403.1876813.1573907695705.ref@mail.yahoo.com> Message-ID: Best option to use hangup_hook On Sat, 16 Nov 2019, 5:58 pm H Yavari via FreeSWITCH-users, < freeswitch-users at lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: H Yavari > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 16 Nov 2019 12:34:55 +0000 (UTC) > Subject: Using Curl in dial plan > Hi to all, > > I am going to use curl and post some info like CID after call hang up to > another web server. Using the Lua and working with events is the only > solution or we have access to call variables in the dial plan when call is > finished or voicemail is done? > > Some useful hints would be appreciated. > > Regards. > H.Yavari > > > > ---------- Forwarded message ---------- > From: H Yavari via FreeSWITCH-users > > To: FreeSWITCH Users Help > Cc: > Bcc: > Date: Sat, 16 Nov 2019 04:58:37 -0800 (PST) > Subject: [Freeswitch-users] Using Curl in dial plan > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 hyavari at rocketmail.com Sat Nov 16 14:09:52 2019 From: hyavari at rocketmail.com (H Yavari) Date: Sat, 16 Nov 2019 14:09:52 +0000 (UTC) Subject: [Freeswitch-users] Using Curl in dial plan In-Reply-To: References: <1146766403.1876813.1573907695705.ref@mail.yahoo.com> Message-ID: <1674165017.1922679.1573913392426@mail.yahoo.com> Thanks. But we can't use same thing in the dial plan right? We should define listener on this event?! Regards,H.Yavari On Saturday, November 16, 2019, 4:53:22 PM GMT+3:30, Faisal Hanif wrote: Best option to use hangup_hook On Sat, 16 Nov 2019, 5:58 pm H Yavari via FreeSWITCH-users, wrote: ---------- Forwarded message ---------- From: H Yavari To: FreeSWITCH Users Help Cc:  Bcc:  Date: Sat, 16 Nov 2019 12:34:55 +0000 (UTC) Subject: Using Curl in dial plan Hi to all, I am going to use curl and post some info like CID after call hang up to another web server. Using the Lua and working with events is the only solution or we have access to call variables in the dial plan when call is finished or voicemail is done? Some useful hints would be appreciated. Regards.H.Yavari ---------- Forwarded message ---------- From: H Yavari via FreeSWITCH-users To: FreeSWITCH Users Help Cc:  Bcc:  Date: Sat, 16 Nov 2019 04:58:37 -0800 (PST) Subject: [Freeswitch-users] Using Curl in dial plan _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users 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 hyavari at rocketmail.com Sat Nov 16 14:37:28 2019 From: hyavari at rocketmail.com (H Yavari) Date: Sat, 16 Nov 2019 14:37:28 +0000 (UTC) Subject: [Freeswitch-users] Using Curl in dial plan In-Reply-To: References: <1146766403.1876813.1573907695705.ref@mail.yahoo.com> Message-ID: <1124051979.1925129.1573915048669@mail.yahoo.com> Yes you are right. I am using the mix of hook and curl. Thank you. On Saturday, November 16, 2019, 4:53:22 PM GMT+3:30, Faisal Hanif wrote: Best option to use hangup_hook On Sat, 16 Nov 2019, 5:58 pm H Yavari via FreeSWITCH-users, wrote: ---------- Forwarded message ---------- From: H Yavari To: FreeSWITCH Users Help Cc:  Bcc:  Date: Sat, 16 Nov 2019 12:34:55 +0000 (UTC) Subject: Using Curl in dial plan Hi to all, I am going to use curl and post some info like CID after call hang up to another web server. Using the Lua and working with events is the only solution or we have access to call variables in the dial plan when call is finished or voicemail is done? Some useful hints would be appreciated. Regards.H.Yavari ---------- Forwarded message ---------- From: H Yavari via FreeSWITCH-users To: FreeSWITCH Users Help Cc:  Bcc:  Date: Sat, 16 Nov 2019 04:58:37 -0800 (PST) Subject: [Freeswitch-users] Using Curl in dial plan _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users 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 hyavari at rocketmail.com Sat Nov 16 14:39:43 2019 From: hyavari at rocketmail.com (H Yavari) Date: Sat, 16 Nov 2019 14:39:43 +0000 (UTC) Subject: [Freeswitch-users] Performance questions In-Reply-To: References: Message-ID: <497776397.1919670.1573915183539@mail.yahoo.com> Jose, do you have any problem with current configurations? On Saturday, November 16, 2019, 4:29:56 PM GMT+3:30, Jose Fco. Irles Durá wrote: Hi, I have several nodes with FreeSWICH, typically 2 socket Xeon v2 (with 8-12 cores every socket) and 64~128GB of RAM. I'm testing different configurations for max performance. I run FreeSWITCH containers (with docker and macvlan network), the base system is Ubuntu 18.04 and I run 4 containers for machine (max concurrency about 500-700 calls and max cps 200, limited in FreeSWITCH config). But I have some questions about the base config of my servers: 1. Is better cpu hyperthreading enabled or disabled? 2. Is relevant NUMA placement for FreeSWITCH (now I bind 2 FreeSWITCH processes to one NUMA node of the machine)? 3. Is it recommended cpu pinning to avoid excessive context switching? (my opinion is yes, but I don't know if the performance difference is too high) 4. I run default Ubuntu kernel (generic ubuntu flavor, with kernel CONFIG_HZ_250=y), but I can change to the "low latency" version, with CONFIG_HZ=1000. Witch is better? I have tested the two versions with timer_test command in FreeSWITCH and I haven't seen any differences (with no load): Avg: 20.000ms Total Time: 1000.002ms Any other advise about network tunning? PD: the containers are Debian Buster with FreeSWITCH 1.10.1 Best regards -- Jose _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users 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 josefu at gmail.com Sat Nov 16 15:30:07 2019 From: josefu at gmail.com (=?UTF-8?Q?Jose_Fco=2E_Irles_Dur=C3=A1?=) Date: Sat, 16 Nov 2019 16:30:07 +0100 Subject: [Freeswitch-users] Performance questions In-Reply-To: References: Message-ID: > Jose, do you have any problem with current configurations? Some users reports audio problems: - some robotic calls - some calls, after a long time talking ok, listen robotic I'm finding possible causes, but the 95% of users that reports problems use Grandstream and I don't if the problem is mine or the phone. I also testing my own network infrastructure and I'm making heuristic captures for check quality. The machine with more load (2 sockets, 16 cores, 32 threads), has about 30% cpu usage at peak hours and 300k context switches. The normal cpu usage in peak hours has about 10% in the other servers Best regards -- Jose From tanim at surroundapps.com Mon Nov 18 02:39:44 2019 From: tanim at surroundapps.com (Md,Mehedi Hasan Kabir(Tanim)) Date: Mon, 18 Nov 2019 08:39:44 +0600 Subject: [Freeswitch-users] Unanswered pstn outbound call via gateway detection In-Reply-To: References: Message-ID: Hi David Thanks for your suggestion. Regards Tanim On Sat, Nov 16, 2019 at 1:12 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > This should go a long way > > > https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=1048914#content/view/1048916 > > > On Fri, 15 Nov 2019 at 19:01, Md,Mehedi Hasan Kabir(Tanim) < > tanim at surroundapps.com> wrote: > >> Hi Ciprian >> >> Thanks for your prompt response. As i am new to freeswitch,can you give >> some guideline/sample about how i can monitor status of the originated >> outbound channel? >> >> Thanks >> Tanim >> >> On Fri, Nov 15, 2019, 10:24 PM Ciprian Dosoftei < >> ciprian.dosoftei at gmail.com> wrote: >> >>> Tanim -- >>> >>> Since you are using ESL, you should monitor the status of the outbound >>> channel you've originated -- you should do that by channel ID, either the >>> generated one or you could preset it via the origination_uuid variable. >>> >>> During the channel's execution you will see a bunch of CHANNEL_* events >>> streamed via ESL; the relevant event for your question is CHANNEL_HANGUP. >>> You can extract the reason behind the hangup from the Hangup-Cause header >>> (could be CALL_REJECTED, USER_BUSY etc. here's a comprehensive list >>> https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table >>> ). >>> >>> Conversely, if the call is picked up, you will see a CHANNEL_ANSWER >>> event coming through. >>> >>> >>> On Fri, 15 Nov 2019 at 10:59, Md,Mehedi Hasan Kabir(Tanim) < >>> tanim at surroundapps.com> wrote: >>> >>>> Hi >>>> >>>> I have originated outbound call to pstn number using gateway provider >>>> flowroute with originate command. I am using python ESL to achieve this. >>>> >>>> If the call is not answered by pstn user, then i need to post status to >>>> my webserver . >>>> >>>> Can anyone suggests me how i can detect whether the pstn call is >>>> answered or not? >>>> >>>> Thanks >>>> Tanim >>>> >>>> _________________________________________________________________________ >>>> >>>> The FreeSWITCH project is sponsored by SignalWire >>>> https://signalwire.com >>>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>>> services. >>>> Build your next product on our scalable cloud platform. >>>> >>>> Join our online community to chat in real time >>>> https://signalwire.community >>>> >>>> Professional FreeSWITCH Services >>>> sales at freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> https://freeswitch.com >>> >>> >>> >>> -- >>> Best Regards, >>> Ciprian Dosoftei >>> >>> The information transmitted is intended only for the addressee and may >>> contain privileged and/or confidential material. If you are not the >>> intended recipient, kindly contact the sender and delete the message. >>> >>> Any disclosure, distribution or copying of this message is strictly >>> prohibited without the expressed permission of the sender. >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users 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 abelcubano at gmail.com Fri Nov 15 22:36:18 2019 From: abelcubano at gmail.com (Abel Monzon) Date: Fri, 15 Nov 2019 17:36:18 -0500 Subject: [Freeswitch-users] Run Bash script in Dialplan Message-ID: Hello. I am trying to run a bash script when dialplan is generate. I have tested this bash script in CLI and is working as expected, but when is called from dialplan I don't see any error and looks like the bash was never runned. api = freeswitch.API(); rotation = api:executeString("system sh /opt/ASTPP/freeswitch/scripts/rotation.sh"); --> this rotate between 2x gateway name and save the name to a file, but looks like never run. gateway = api:executeString("system sh /opt/ASTPP/freeswitch/scripts/random2.sh"); ---> this script read the name of the gateway from the file, and works perfect. table.insert(xml, [[]]); Here is the result on console: EXECUTE sofia/default/14254358039 at 198.50.167.93 bridge([leg_timeout=0,absolute_codec_string=^^:PCMA:PCMU:G729]sofia/gateway/GATEWAY-01/38227043) I don't any way to troubleshoot why 1x script run perfectly and the other one is not running. Thank You in advance. Abel -------------- next part -------------- An HTML attachment was scrubbed... URL: From gaurang.gohil at ecosmob.com Mon Nov 18 08:34:47 2019 From: gaurang.gohil at ecosmob.com (Gaurang Gohil) Date: Mon, 18 Nov 2019 14:04:47 +0530 Subject: [Freeswitch-users] Fwd: contact missing in 180 response In-Reply-To: References: Message-ID: ---------- Forwarded message --------- From: Gaurang Gohil Date: Thu, Nov 7, 2019 at 6:40 PM Subject: contact missing in 180 response To: Hello all, I set up a presence in linphone SDK and have to face the problem of contact loss in Sip Dialog cause of linephone SDK not add contact header in 180 or 183 packets, I tried to debug and found that in Sofia.c when we insert dialog in sip dialog table at the time of 180 or 183 with state of "early" but linephone does not add CONTACT header in 180 packet so Sofia unable to get contact header in 180 packet and due to this we lose the presence state when FS receive subscribe and unable to get contact in sip dialog table, please help me on this. please find the pcap i have attached to this email. -- Regards, *Gaurang Gohil* | Jr. Software Developer +91 9662008820 | Hangout: gaurang.gohil at ecosmob.com [image: Ecosmob Technologies Pvt. Ltd.] Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data ssdsds sasadsdasdasdasdasdas This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- Regards, *Gaurang Gohil* | Jr. Software Developer +91 9662008820 | Hangout: gaurang.gohil at ecosmob.com [image: Ecosmob Technologies Pvt. Ltd.] Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data ssdsds sasadsdasdasdasdasdas This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: IMG_20191107_164014.jpg Type: image/jpeg Size: 81019 bytes Desc: not available URL: From Paul.Mateer at outlook.com Mon Nov 18 16:48:34 2019 From: Paul.Mateer at outlook.com (Paul Mateer) Date: Mon, 18 Nov 2019 16:48:34 +0000 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. Message-ID: Hi. I have a FreeSWITCH system up and running and I can make calls from one device to another which is fine. What I would now like to do is integrate it with another system so that when certain events occur in this other system, it dials one or more destinations and plays a pre-recorded message before hanging up. Can the playing of the sound file be hooked into a custom dialplan? I ask because a little experimentation has revealed that the dialplan is processed on the inbound channel (which seems entirely reasonable) which means that using playback in the dialplan results in audio sent to the caller, rather than the called. If this can't be achieved via the dialplan, does this mean that I need to create a specific client that can call a destination and somehow pipe sound into the input channel, rather than taking it from whatever input device might be available? Thanks for any assistance offered. Paul Sent from my Windows 10 device -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Mon Nov 18 19:44:35 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Mon, 18 Nov 2019 14:44:35 -0500 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. In-Reply-To: References: Message-ID: It is absolutely feasible to implement your use case via the dialplan; check the first example here: https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example If the use case is as simple as placing an outbound call, playing back some media and then hanging up, you could implement is as: originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message) More information here: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate On Mon, 18 Nov 2019 at 14:33, Paul Mateer wrote: > Hi. I have a FreeSWITCH system up and running and I can make calls from > one device to another which is fine. > > > > What I would now like to do is integrate it with another system so that > when certain events occur in this other system, it dials one or more > destinations and plays a pre-recorded message before hanging up. > > > > Can the playing of the sound file be hooked into a custom dialplan? I ask > because a little experimentation has revealed that the dialplan is > processed on the inbound channel (which seems entirely reasonable) which > means that using playback in the dialplan results in audio sent to the > caller, rather than the called. > > > > If this can't be achieved via the dialplan, does this mean that I need to > create a specific client that can call a destination and somehow pipe sound > into the input channel, rather than taking it from whatever input device > might be available? > > > > Thanks for any assistance offered. > > > > Paul > > > > > > Sent from my Windows 10 device > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From davidswalkabout at gmail.com Mon Nov 18 23:16:15 2019 From: davidswalkabout at gmail.com (David P) Date: Mon, 18 Nov 2019 15:16:15 -0800 Subject: [Freeswitch-users] Private IP used for audio even though ACL should block it Message-ID: In the log snippet below, a private IP is chosen for the audio connection (6th line from bottom) even though our acl.conf.xml should reject it. How can we be sure our ACL is enforced? Our version on debian9: FreeSWITCH Version 20.19.4-release-12-fc9d51c~64bit (-release-12-fc9d51c 64bit) Our autoload_configs/acl.conf.xml contains: And our verto.conf.xml's "default-v4" profile references it: (Also, strangely, we get a relay candidate from our TURN server for video but not audio.) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e m=audio 57658 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 110 113 126 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e c=IN IP4 10.0.0.90 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=rtcp:9 IN IP4 0.0.0.0 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2964267771 1 udp 2113937151 10.0.0.90 57658 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2509878665 1 udp 2113939711 2601:ipv6:address 57659 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e m=video 12769 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 125 104 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e c=IN IP4 52.public.turn.addr 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=rtcp:9 IN IP4 0.0.0.0 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2964267771 1 udp 2113937151 10.0.0.90 52903 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2509878665 1 udp 2113939711 2601:ipv6:address 52904 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2563966249 1 udp 16785151 52.public.turn.addr 12769 typ relay raddr 73.user.public.ip rport 49605 generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_codec.c:111 verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 Original read codec set to opus:116 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.0.90:57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4227 Drop audio Candidate cid: 1 proto: udp type: host addr: 2601:ipv6:address:57659 (no network path) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_vpx.c:703 VPX VER:v1.7.0 VPX_IMAGE_ABI_VERSION:4 VPX_CODEC_ABI_VERSION:8 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_vpx.c:703 VPX VER:v1.7.0 VPX_IMAGE_ABI_VERSION:4 VPX_CODEC_ABI_VERSION:8 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:3598 Set VIDEO Codec verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 VP8/90000 0 ms 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save video Candidate cid: 1 proto: udp type: host addr: 10.0.0.90:52903 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4227 Drop video Candidate cid: 1 proto: udp type: host addr: 2601:ipv6:address:52904 (no network path) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save video Candidate cid: 1 proto: udp type: relay addr: 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtcp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4325 Look for Relay Candidates as last resort 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4287 Choose rtp candidate, index 1, 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4053 verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 choosing family v4 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4298 Choose same candidate, index 0, for rtcp based on rtcp-mux attribute 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4350 setting remote video ice addr to index 1 52.public.turn.addr:12769 based on candidate 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4385 Setting remote rtcp video addr to 52.public.turn.addr:12769 based on candidate 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:8600 AUDIO RTP [verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1] 10.0.0.192 port 26664 -> 10.0.0.90 port 57658 codec: 111 ms: 20 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_rtp.c:4480 Starting timer [soft] 960 bytes per 20ms 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:8819 Activating RTCP PORT 57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_rtp.c:4880 RTCP send rate is: 1000 and packet rate is: 20000 Remote Port: 57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [INFO] switch_rtp.c:3808 Activate RTP/RTCP audio DTLS client 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [INFO] switch_rtp.c:3975 Changing audio DTLS state from OFF to HANDSHAKE -------------- next part -------------- An HTML attachment was scrubbed... URL: From Paul.Mateer at outlook.com Tue Nov 19 10:22:05 2019 From: Paul.Mateer at outlook.com (Paul Mateer) Date: Tue, 19 Nov 2019 10:22:05 +0000 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. In-Reply-To: References: , Message-ID: I must be doing something wrong then, because I run a command of the form originate XML public &playback(c:/temp/test.wav) and the call goes through, but I don't hear anything at the called end. I do see an entry in the log like: Transfer sofia/internal/gateway at ip:port to public[xml@&playback(c:/temp/test.wav)] Which presumably was caused by the playback command, but I don't know if this is to be expected, and i it is whether the details are correct. Paul Sent from my Windows 10 device From: Ciprian Dosoftei Sent: 18 November 2019 20:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling a destination and playing a pre-recorded message. It is absolutely feasible to implement your use case via the dialplan; check the first example here: https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example If the use case is as simple as placing an outbound call, playing back some media and then hanging up, you could implement is as: originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message) More information here: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate On Mon, 18 Nov 2019 at 14:33, Paul Mateer > wrote: Hi. I have a FreeSWITCH system up and running and I can make calls from one device to another which is fine. What I would now like to do is integrate it with another system so that when certain events occur in this other system, it dials one or more destinations and plays a pre-recorded message before hanging up. Can the playing of the sound file be hooked into a custom dialplan? I ask because a little experimentation has revealed that the dialplan is processed on the inbound channel (which seems entirely reasonable) which means that using playback in the dialplan results in audio sent to the caller, rather than the called. If this can't be achieved via the dialplan, does this mean that I need to create a specific client that can call a destination and somehow pipe sound into the input channel, rather than taking it from whatever input device might be available? Thanks for any assistance offered. Paul Sent from my Windows 10 device _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 19 12:07:12 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 19 Nov 2019 12:07:12 +0000 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. In-Reply-To: References: Message-ID: Hello, You need to put the file you want to play in a directory fs can access and not pass anything like "c:\". I store them (i only need them temporarily) in /tmp/ I.e.: bgapi originate {origination_caller_id_number=[CALLER-ID}sofia/gateway/nexvortex/[SOME-DESTINATION-NUMBER] &playback(/tmp/test.wav) Note the dialstring. Hope that helps Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Nov 19, 2019 at 10:36 AM Paul Mateer wrote: > I must be doing something wrong then, because I run a command of the form > > > > originate XML public &playback(c:/temp/test.wav) > > > > and the call goes through, but I don't hear anything at the called end. I > do see an entry in the log like: > > > > Transfer sofia/internal/gateway at ip:port to public[xml@ > &playback(c:/temp/test.wav)] > > > > Which presumably was caused by the playback command, but I don't know if > this is to be expected, and i it is whether the details are correct. > > > > Paul > > > > Sent from my Windows 10 device > > > > *From: *Ciprian Dosoftei > *Sent: *18 November 2019 20:01 > *To: *FreeSWITCH Users Help > *Subject: *Re: [Freeswitch-users] Calling a destination and playing a > pre-recorded message. > > > It is absolutely feasible to implement your use case via the dialplan; > check the first example here: > > https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example > > If the use case is as simple as placing an outbound call, playing back > some media and then hanging up, you could implement is as: > > originate > {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number > &playback(message) > > More information here: > https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate > > On Mon, 18 Nov 2019 at 14:33, Paul Mateer wrote: > >> Hi. I have a FreeSWITCH system up and running and I can make calls from >> one device to another which is fine. >> >> >> >> What I would now like to do is integrate it with another system so that >> when certain events occur in this other system, it dials one or more >> destinations and plays a pre-recorded message before hanging up. >> >> >> >> Can the playing of the sound file be hooked into a custom dialplan? I ask >> because a little experimentation has revealed that the dialplan is >> processed on the inbound channel (which seems entirely reasonable) which >> means that using playback in the dialplan results in audio sent to the >> caller, rather than the called. >> >> >> >> If this can't be achieved via the dialplan, does this mean that I need to >> create a specific client that can call a destination and somehow pipe sound >> into the input channel, rather than taking it from whatever input device >> might be available? >> >> >> >> Thanks for any assistance offered. >> >> >> >> Paul >> >> >> >> >> >> Sent from my Windows 10 device >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 19 12:08:51 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 19 Nov 2019 12:08:51 +0000 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. In-Reply-To: References: Message-ID: BTW, yes you can use as a dialstring something like bgapi originate sofia/internal/[DEST-NUMBER]@IP:PORT &playback(/tmp/test.wav) Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Nov 19, 2019 at 12:07 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello, > > You need to put the file you want to play in a directory fs can access > and not pass anything like "c:\". I store them (i only need them > temporarily) in /tmp/ > > I.e.: > > bgapi originate > {origination_caller_id_number=[CALLER-ID}sofia/gateway/nexvortex/[SOME-DESTINATION-NUMBER] > &playback(/tmp/test.wav) > > Note the dialstring. > > Hope that helps > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Tue, Nov 19, 2019 at 10:36 AM Paul Mateer > wrote: > >> I must be doing something wrong then, because I run a command of the form >> >> >> >> originate XML public &playback(c:/temp/test.wav) >> >> >> >> and the call goes through, but I don't hear anything at the called end. I >> do see an entry in the log like: >> >> >> >> Transfer sofia/internal/gateway at ip:port to public[xml@ >> &playback(c:/temp/test.wav)] >> >> >> >> Which presumably was caused by the playback command, but I don't know if >> this is to be expected, and i it is whether the details are correct. >> >> >> >> Paul >> >> >> >> Sent from my Windows 10 device >> >> >> >> *From: *Ciprian Dosoftei >> *Sent: *18 November 2019 20:01 >> *To: *FreeSWITCH Users Help >> *Subject: *Re: [Freeswitch-users] Calling a destination and playing a >> pre-recorded message. >> >> >> It is absolutely feasible to implement your use case via the dialplan; >> check the first example here: >> >> https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example >> >> If the use case is as simple as placing an outbound call, playing back >> some media and then hanging up, you could implement is as: >> >> originate >> {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number >> &playback(message) >> >> More information here: >> https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate >> >> On Mon, 18 Nov 2019 at 14:33, Paul Mateer >> wrote: >> >>> Hi. I have a FreeSWITCH system up and running and I can make calls from >>> one device to another which is fine. >>> >>> >>> >>> What I would now like to do is integrate it with another system so that >>> when certain events occur in this other system, it dials one or more >>> destinations and plays a pre-recorded message before hanging up. >>> >>> >>> >>> Can the playing of the sound file be hooked into a custom dialplan? I >>> ask because a little experimentation has revealed that the dialplan is >>> processed on the inbound channel (which seems entirely reasonable) which >>> means that using playback in the dialplan results in audio sent to the >>> caller, rather than the called. >>> >>> >>> >>> If this can't be achieved via the dialplan, does this mean that I need >>> to create a specific client that can call a destination and somehow pipe >>> sound into the input channel, rather than taking it from whatever input >>> device might be available? >>> >>> >>> >>> Thanks for any assistance offered. >>> >>> >>> >>> Paul >>> >>> >>> >>> >>> >>> Sent from my Windows 10 device >>> >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Best Regards, >> Ciprian Dosoftei >> >> The information transmitted is intended only for the addressee and may >> contain privileged and/or confidential material. If you are not the >> intended recipient, kindly contact the sender and delete the message. >> >> Any disclosure, distribution or copying of this message is strictly >> prohibited without the expressed permission of the sender. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Paul.Mateer at outlook.com Tue Nov 19 14:46:09 2019 From: Paul.Mateer at outlook.com (Paul Mateer) Date: Tue, 19 Nov 2019 14:46:09 +0000 Subject: [Freeswitch-users] Calling a destination and playing a pre-recorded message. In-Reply-To: References: , Message-ID: Great. Finally got things working. Thanks for the tip on keeping the sound files within the FreeSWITCH hierarchy. Paul Sent from my Windows 10 device From: David Villasmil Sent: 19 November 2019 13:00 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling a destination and playing a pre-recorded message. Hello, You need to put the file you want to play in a directory fs can access and not pass anything like "c:\". I store them (i only need them temporarily) in /tmp/ I.e.: bgapi originate {origination_caller_id_number=[CALLER-ID}sofia/gateway/nexvortex/[SOME-DESTINATION-NUMBER] &playback(/tmp/test.wav) Note the dialstring. Hope that helps Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Nov 19, 2019 at 10:36 AM Paul Mateer > wrote: I must be doing something wrong then, because I run a command of the form originate XML public &playback(c:/temp/test.wav) and the call goes through, but I don't hear anything at the called end. I do see an entry in the log like: Transfer sofia/internal/gateway at ip:port to public[xml@&playback(c:/temp/test.wav)] Which presumably was caused by the playback command, but I don't know if this is to be expected, and i it is whether the details are correct. Paul Sent from my Windows 10 device From: Ciprian Dosoftei Sent: 18 November 2019 20:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling a destination and playing a pre-recorded message. It is absolutely feasible to implement your use case via the dialplan; check the first example here: https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example If the use case is as simple as placing an outbound call, playing back some media and then hanging up, you could implement is as: originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message) More information here: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-originate On Mon, 18 Nov 2019 at 14:33, Paul Mateer > wrote: Hi. I have a FreeSWITCH system up and running and I can make calls from one device to another which is fine. What I would now like to do is integrate it with another system so that when certain events occur in this other system, it dials one or more destinations and plays a pre-recorded message before hanging up. Can the playing of the sound file be hooked into a custom dialplan? I ask because a little experimentation has revealed that the dialplan is processed on the inbound channel (which seems entirely reasonable) which means that using playback in the dialplan results in audio sent to the caller, rather than the called. If this can't be achieved via the dialplan, does this mean that I need to create a specific client that can call a destination and somehow pipe sound into the input channel, rather than taking it from whatever input device might be available? Thanks for any assistance offered. Paul Sent from my Windows 10 device _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales at freeswitch.com https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Nov 21 18:11:26 2019 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 21 Nov 2019 18:11:26 +0000 Subject: [Freeswitch-users] Failed DTMF sanity check In-Reply-To: References: Message-ID: You can try FreeSwitch version 1.10.1 ( https://github.com/signalwire/freeswitch/releases) . However, that looks like an issue with your provider. On Mon, Nov 11, 2019 at 9:13 PM Saqib Nawaz wrote: > Thanks for your reply, I'm using FreeSWITCH version: 1.6.9. No idea of the > device connected to other ends. > > On Tue, 12 Nov 2019 at 01:26, Ciprian Dosoftei > wrote: > >> It typically means the device on the other end of the channel is sending >> in malformed RFC2833 DTMF. >> >> What FreeSWITCH version are you using, and also, do you know what kind of >> device is connected on the other end? >> >> On Mon, 11 Nov 2019 at 14:28, Saqib Nawaz >> wrote: >> >>> Hi Guys, >>> >>> I'm getting this error message printed on my console >>> >>> [ERR] switch_rtp.c:2013 Failed DTMF sanity check. >>> >>> Why this is popping up ? how can I prevent this from occurring? >>> >>> Regards, >>> Saqib Nawaz. >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> https://freeswitch.com >> >> >> >> -- >> Best Regards, >> Ciprian Dosoftei >> >> The information transmitted is intended only for the addressee and may >> contain privileged and/or confidential material. If you are not the >> intended recipient, kindly contact the sender and delete the message. >> >> Any disclosure, distribution or copying of this message is strictly >> prohibited without the expressed permission of the sender. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From shahnawaj.khan1990 at gmail.com Mon Nov 25 13:39:26 2019 From: shahnawaj.khan1990 at gmail.com (Shahnawaj Khan) Date: Mon, 25 Nov 2019 19:09:26 +0530 Subject: [Freeswitch-users] FS firing Curl request for extension transfer Message-ID: Hi, I am using mod_xml_curl to fetch dialplan dynamically. On the Basis of condition I need to jump from one extension to another within the same context. my dialplan is similar to the below example. In above case FS is firing curl request for every extension jump within context. Is there any way to jump from one extension to another within same context without firing new request. I don't want to use execute_extension as it return right back to where it was called from. Thanks & Regards, Shahnawaz From brian at freeswitch.com Mon Nov 25 14:13:04 2019 From: brian at freeswitch.com (Brian West) Date: Mon, 25 Nov 2019 08:13:04 -0600 Subject: [Freeswitch-users] FS firing Curl request for extension transfer In-Reply-To: References: Message-ID: No, if you're thinking along those lines you're doing it wrong, it will fire a request on every transfer, or dip into the dialplan, You should never ever return a dialplan with an expression in the condition field, you already know the answer, just give freeswitch an empty condition with the exact steps you wish it to perform. DO NOT give it more than a single extension, its a waste of time otherwise. /b On Mon, Nov 25, 2019 at 7:58 AM Shahnawaj Khan wrote: > Hi, > > I am using mod_xml_curl to fetch dialplan dynamically. On the Basis of > condition I need to jump from one extension to another within the same > context. my dialplan is similar to the below example. > > > > > > > > > > > > > > > > > In above case FS is firing curl request for every extension jump > within context. Is there any way to jump from one extension to another > within same context without firing new request. I don't want to use > execute_extension as it return right back to where it was called from. > > Thanks & Regards, > Shahnawaz > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From ayoub.ichchou at airenetworks.es Fri Nov 22 13:19:29 2019 From: ayoub.ichchou at airenetworks.es (Ayoub Ichchou) Date: Fri, 22 Nov 2019 13:19:29 +0000 Subject: [Freeswitch-users] FreeSwitch database need help In-Reply-To: References: <4e0fcaf8aa8949cfbc60c1fa0054c523@airenetworks.es> Message-ID: Good afternoon, I have been using FreeSwitch for a long time and so far it works like silk. The thing is that now I have engaged in a new adventure with FS, i want to use FS as SBC. For this I want to mount FS with a MySQL database and then form a cluster. I found the problem when I try to mount the database with FS. I am sure that there is a way for FS to generate the database, the necessary tables and fill them with the necessary data to manage FS as SBC, but I cannot find documentation about it. I have the FS books and I have read the confluence wiki. I want FS to generate the necessary database and tables because I don't know how to create the database data for FS to use to make all the SBC functionality. Where can I find the documentation for it? what are the steps to create the database to use FS as SBC? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: From ruslan at shevchenko.kiev.ua Fri Nov 22 07:50:29 2019 From: ruslan at shevchenko.kiev.ua (Ruslan Shevchenko) Date: Fri, 22 Nov 2019 09:50:29 +0200 Subject: [Freeswitch-users] bgsystem call Message-ID: Hi, I try to use 'bgsystem' call from mod_dptools API. With the next two entries in dialplan, I guess, the session should wait until calling party stop call: But looks, like the session, endsed immediately after the process, spawned by bgsystem, finished (and before we receive result). Also, looks like the help page on the wiki (about the possibility to return data from bgsystem: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bgsystem) is incorrect, it is impossible to receive output in channel variable, as I can understand. Is handling session is expected behavior ? Maybe exists some way to handle events from bgprocess ? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdirahman.osm at gmail.com Thu Nov 21 14:27:12 2019 From: abdirahman.osm at gmail.com (Abdirahman A. Osman) Date: Thu, 21 Nov 2019 09:27:12 -0500 Subject: [Freeswitch-users] CentOS 7 systemctl start freeswitch Message-ID: Hi, I am using CentOS 7 and tried to start the freeswitch with systemctl start freeswitch. It is working with freeswitch -nc but the systemctl start freeswitch is using the freeswitch:daemon user, a shown below root at localhost bin]# systemctl status freeswitch.service ● freeswitch.service - freeswitch Loaded: loaded (/usr/lib/systemd/system/freeswitch.service; enabled; vendor preset: disabled) Active: failed (Result: start-limit) since Thu 2019-11-21 09:05:16 EST; 8s ago Process: 4251 ExecStart=/usr/bin/freeswitch -u freeswitch -g daemon -ncwait $DAEMON_OPTS (code=exited, status=1/FAILURE) Process: 4249 ExecStartPre=/bin/chown -R freeswitch:daemon /var/run/freeswitch/ (code=exited, status=0/SUCCESS) Process: 4248 ExecStartPre=/bin/mkdir -p /var/run/freeswitch/ (code=exited, status=0/SUCCESS) Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service: control process exited, code=exited status=1 Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start freeswitch. Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service entered failed state. Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service failed. Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service holdoff time over, scheduling restart. Nov 21 09:05:16 localhost.localdomain systemd[1]: Stopped freeswitch. Nov 21 09:05:16 localhost.localdomain systemd[1]: start request repeated too quickly for freeswitch.service Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start freeswitch. Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service entered failed state. Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service failed. Although permissions of the folders the below folders are set for the user freeswitch drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ drwxr-x--- 2 freeswitch daemon 40 Nov 21 09:05 /var/run/freeswitch/ drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ -rwxr-xr-x. 1 freeswitch daemon 28336 Aug 20 15:08 /usr/bin/freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Mon Nov 25 19:56:21 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Mon, 25 Nov 2019 14:56:21 -0500 Subject: [Freeswitch-users] CentOS 7 systemctl start freeswitch In-Reply-To: References: Message-ID: Check the permissions for /var/log/freeswitch/ too. If it's fine, look at the data collected in /var/log/freeswitch/freeswitch.log; towards the end of the file you should have some meaningful explication for the ungraceful/premature exit. On Mon, 25 Nov 2019 at 14:07, Abdirahman A. Osman wrote: > Hi, > > I am using CentOS 7 and tried to start the freeswitch with systemctl > start freeswitch. > > It is working with freeswitch -nc but the systemctl start freeswitch is > using the freeswitch:daemon user, a shown below > > > root at localhost bin]# systemctl status freeswitch.service > ● freeswitch.service - freeswitch > Loaded: loaded (/usr/lib/systemd/system/freeswitch.service; enabled; > vendor preset: disabled) > Active: failed (Result: start-limit) since Thu 2019-11-21 09:05:16 EST; > 8s ago > Process: 4251 ExecStart=/usr/bin/freeswitch -u freeswitch -g daemon > -ncwait $DAEMON_OPTS (code=exited, status=1/FAILURE) > Process: 4249 ExecStartPre=/bin/chown -R freeswitch:daemon > /var/run/freeswitch/ (code=exited, status=0/SUCCESS) > Process: 4248 ExecStartPre=/bin/mkdir -p /var/run/freeswitch/ > (code=exited, status=0/SUCCESS) > > Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service: > control process exited, code=exited status=1 > Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start > freeswitch. > Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service > entered failed state. > Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service > failed. > Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service > holdoff time over, scheduling restart. > Nov 21 09:05:16 localhost.localdomain systemd[1]: Stopped freeswitch. > Nov 21 09:05:16 localhost.localdomain systemd[1]: start request repeated > too quickly for freeswitch.service > Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start > freeswitch. > Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service > entered failed state. > Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service > failed. > > > Although permissions of the folders the below folders are set for the user > freeswitch > > drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ > drwxr-x--- 2 freeswitch daemon 40 Nov 21 09:05 /var/run/freeswitch/ > drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ > -rwxr-xr-x. 1 freeswitch daemon 28336 Aug 20 15:08 /usr/bin/freeswitch > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jprangi at gmail.com Mon Nov 25 20:22:11 2019 From: jprangi at gmail.com (Jai Rangi) Date: Mon, 25 Nov 2019 12:22:11 -0800 Subject: [Freeswitch-users] Delay in Options Packets Message-ID: Hello, Every now and then we see long delay in response for options. Example in the trace below free-switch took 44 seconds before it send 200 OK. What will be best way to troubleshoot this? U 2019/11/25 10:49:27.185325 192.168.1.1:49020 -> 192.168.1.5:5060 OPTIONS sip:100000 at 192.168.1.5 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.1:49020;branch=z9hG4bKhjhs8ass877. Max-Forwards: 70. To: sip:100000 at 192.168.1.5. From: sip:1000 at 192.168.2.1:49020;tag=cbzxq0. Call-ID: hcrtfi at 192.168.2.1. CSeq: 1 OPTIONS. Accept: application/sdp. Content-Length: 0. . U 2019/11/25 10:49:51.819959 192.168.1.5:5060 -> 192.168.1.1:49020 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.1:49020 ;branch=z9hG4bKhjhs8ass877;received=192.168.1.1. From: ;tag=cbzxq0. To: ;tag=aUUQj55rcetma. Call-ID: hcrtfi at 192.168.2.1. CSeq: 1 OPTIONS. Contact: . User-Agent: Monitor. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: path, replaces. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 25 21:54:31 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 25 Nov 2019 21:54:31 +0000 Subject: [Freeswitch-users] CentOS 7 systemctl start freeswitch In-Reply-To: References: Message-ID: Start it from the command line as freeswitch. this is very probably a permissions issue as Ciprian pointed out. How did you install FS? Did you compile it or installed from some package? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Mon, Nov 25, 2019 at 8:24 PM Ciprian Dosoftei wrote: > Check the permissions for /var/log/freeswitch/ too. If it's fine, look at > the data collected in /var/log/freeswitch/freeswitch.log; towards the end > of the file you should have some meaningful explication for the > ungraceful/premature exit. > > On Mon, 25 Nov 2019 at 14:07, Abdirahman A. Osman < > abdirahman.osm at gmail.com> wrote: > >> Hi, >> >> I am using CentOS 7 and tried to start the freeswitch with systemctl >> start freeswitch. >> >> It is working with freeswitch -nc but the systemctl start freeswitch is >> using the freeswitch:daemon user, a shown below >> >> >> root at localhost bin]# systemctl status freeswitch.service >> ● freeswitch.service - freeswitch >> Loaded: loaded (/usr/lib/systemd/system/freeswitch.service; enabled; >> vendor preset: disabled) >> Active: failed (Result: start-limit) since Thu 2019-11-21 09:05:16 >> EST; 8s ago >> Process: 4251 ExecStart=/usr/bin/freeswitch -u freeswitch -g daemon >> -ncwait $DAEMON_OPTS (code=exited, status=1/FAILURE) >> Process: 4249 ExecStartPre=/bin/chown -R freeswitch:daemon >> /var/run/freeswitch/ (code=exited, status=0/SUCCESS) >> Process: 4248 ExecStartPre=/bin/mkdir -p /var/run/freeswitch/ >> (code=exited, status=0/SUCCESS) >> >> Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service: >> control process exited, code=exited status=1 >> Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start >> freeswitch. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service >> entered failed state. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service >> failed. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service >> holdoff time over, scheduling restart. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: Stopped freeswitch. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: start request repeated >> too quickly for freeswitch.service >> Nov 21 09:05:16 localhost.localdomain systemd[1]: Failed to start >> freeswitch. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: Unit freeswitch.service >> entered failed state. >> Nov 21 09:05:16 localhost.localdomain systemd[1]: freeswitch.service >> failed. >> >> >> Although permissions of the folders the below folders are set for the >> user freeswitch >> >> drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ >> drwxr-x--- 2 freeswitch daemon 40 Nov 21 09:05 /var/run/freeswitch/ >> drwxr-xr-x 13 freeswitch daemon 4096 Nov 17 15:36 /etc/freeswitch/ >> -rwxr-xr-x. 1 freeswitch daemon 28336 Aug 20 15:08 /usr/bin/freeswitch >> >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Best Regards, > Ciprian Dosoftei > > The information transmitted is intended only for the addressee and may > contain privileged and/or confidential material. If you are not the > intended recipient, kindly contact the sender and delete the message. > > Any disclosure, distribution or copying of this message is strictly > prohibited without the expressed permission of the sender. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Nov 25 22:18:00 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 25 Nov 2019 22:18:00 +0000 Subject: [Freeswitch-users] FreeSwitch database need help In-Reply-To: References: <4e0fcaf8aa8949cfbc60c1fa0054c523@airenetworks.es> Message-ID: You don’t need to, fs will create all the tables it needs of it can’t find them. Just connect fs to a db in a schema and it will create them all. If you’re on the 1.10 you must enable mod_sql or mod_psql and then configure the dan in the core using a proper connection string. This is outdated for 1.10 but it points in the right direction. https://freeswitch.org/confluence/plugins/servlet/mobile?contentId=6587804#content/view/6587804 Hope that helps, David On Mon, 25 Nov 2019 at 19:22, Ayoub Ichchou wrote: > Good afternoon, > > > > I have been using FreeSwitch for a long time and so far it works like > silk. The thing is that now I have engaged in a new adventure with FS, i > want to use FS as SBC. For this I want to mount FS with a MySQL database > and then form a cluster. > > > > I found the problem when I try to mount the database with FS. I am sure > that there is a way for FS to generate the database, the necessary tables > and fill them with the necessary data to manage FS as SBC, but I cannot > find documentation about it. I have the FS books and I have read the > confluence wiki. I want FS to generate the necessary database and tables > because I don't know how to create the database data for FS to use to make > all the SBC functionality. Where can I find the documentation for it? what > are the steps to create the database to use FS as SBC? > > > > Regards. > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at freeswitch.org Tue Nov 26 00:32:14 2019 From: mike at freeswitch.org (Mike Jerris) Date: Mon, 25 Nov 2019 16:32:14 -0800 Subject: [Freeswitch-users] Delay in Options Packets In-Reply-To: References: Message-ID: <0DD51286-4E72-4159-AB48-51FC27CDFF76@freeswitch.org> Our code to handle options couldn’t be much simpler. If you are seeing massive delays the only thing I would guess would be that there is some sort of massive ddos type attack causing significant queuing of inbound requests or a register storm and your registrations handlers are getting behind. Take a look at logging and sip traffic around the same time for clues on what else might be happening on this sip profile at the same time. > On Nov 25, 2019, at 12:22 PM, Jai Rangi wrote: > > Hello, > > Every now and then we see long delay in response for options. Example in the trace below free-switch took 44 seconds before it send 200 OK. > > What will be best way to troubleshoot this? > > U 2019/11/25 10:49:27.185325 192.168.1.1:49020 -> 192.168.1.5:5060 > OPTIONS sip:100000 at 192.168.1.5 SIP/2.0. > Via: SIP/2.0/UDP 192.168.2.1:49020;branch=z9hG4bKhjhs8ass877. > Max-Forwards: 70. > To: sip:100000 at 192.168.1.5 . > From: sip:1000 at 192.168.2.1:49020;tag=cbzxq0. > Call-ID: hcrtfi at 192.168.2.1 . > CSeq: 1 OPTIONS. > Accept: application/sdp. > Content-Length: 0. > . > > U 2019/11/25 10:49:51.819959 192.168.1.5:5060 -> 192.168.1.1:49020 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.2.1:49020;branch=z9hG4bKhjhs8ass877;received=192.168.1.1. > From: >;tag=cbzxq0. > To: >;tag=aUUQj55rcetma. > Call-ID: hcrtfi at 192.168.2.1 . > CSeq: 1 OPTIONS. > Contact: >. > User-Agent: Monitor. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. > Supported: path, replaces. > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. > Content-Length: 0. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.com Tue Nov 26 01:30:46 2019 From: brian at freeswitch.com (Brian West) Date: Mon, 25 Nov 2019 19:30:46 -0600 Subject: [Freeswitch-users] bgsystem call In-Reply-To: References: Message-ID: Not unless you use inline=true probably on the action tag. On Mon, Nov 25, 2019 at 12:53 PM Ruslan Shevchenko < ruslan at shevchenko.kiev.ua> wrote: > Hi, I try to use 'bgsystem' call from mod_dptools API. With the next > two entries in dialplan, I guess, the session should wait until calling > party stop call: > > > > > But looks, like the session, endsed immediately after the process, spawned > by bgsystem, finished (and before we receive result). > > Also, looks like the help page on the wiki (about the possibility to > return data from bgsystem: > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+bgsystem) > is incorrect, it is impossible to receive output in channel variable, as I > can understand. > > Is handling session is expected behavior ? Maybe exists some way to > handle events from bgprocess ? > > 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 -- Brian West | Co-founder and Developer Need Commercial support? email sales at freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 Email: brian at freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com [image: https://www.facebook.com/signalwireinc?src=email] [image: https://twitter.com/freeswitch] -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Nov 26 17:16:11 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 26 Nov 2019 17:16:11 +0000 Subject: [Freeswitch-users] core.db and postgresql Message-ID: Hello all, I configured postgres in the core, tables were created, all good. But i just noticed that the sqlite3 files in /var/lib/freeswitch/db/ are still being updated: -rw-r----- 1 freeswitch freeswitch 14336 Jun 3 18:14 /var/lib/freeswitch/db/call_limit.db -rw-r--r-- 1 freeswitch freeswitch 242688 Nov 26 12:13 /var/lib/freeswitch/db/core.db -rw-r--r-- 1 freeswitch freeswitch 4054016 Nov 26 11:10 /var/lib/freeswitch/db/sofia_reg_external.db -rw-rw---- 1 freeswitch freeswitch 33 Nov 26 12:11 /var/lib/freeswitch/db/zrtp.dat Is this right? Isn't the point of using db in the core _not_ to use sqlite? Thanks for your help! David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Tue Nov 26 17:44:12 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 26 Nov 2019 18:44:12 +0100 Subject: [Freeswitch-users] core.db and postgresql In-Reply-To: References: Message-ID: On Tue, Nov 26, 2019 at 6:41 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > > I configured postgres in the core, tables were created, all good. > But i just noticed that the sqlite3 files in /var/lib/freeswitch/db/ are > still being updated: > > I believe would be better if you stop FS, delete those sqlite db files, then restart FS. > -rw-r----- 1 freeswitch freeswitch 14336 Jun 3 18:14 > /var/lib/freeswitch/db/call_limit.db > -rw-r--r-- 1 freeswitch freeswitch 242688 Nov 26 12:13 > /var/lib/freeswitch/db/core.db > -rw-r--r-- 1 freeswitch freeswitch 4054016 Nov 26 11:10 > /var/lib/freeswitch/db/sofia_reg_external.db > -rw-rw---- 1 freeswitch freeswitch 33 Nov 26 12:11 > /var/lib/freeswitch/db/zrtp.dat > > Is this right? Isn't the point of using db in the core _not_ to use sqlite? > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Tue Nov 26 18:09:13 2019 From: asilva at wirelessmundi.com (=?UTF-8?Q?Ant=c3=b3nio_Silva?=) Date: Tue, 26 Nov 2019 19:09:13 +0100 Subject: [Freeswitch-users] call drop because of no write codec, incompatible_destination In-Reply-To: References: <7C828828-2364-457F-8398-580EEF04BB73@freeswitch.org> Message-ID: <0206eb53-0378-14c7-f683-4b27ca07d695@wirelessmundi.com> Hi, Actually my problem didn't get fixed.. restart  freeswitch it solve the issue but after a few days it start to reproduce again, and not for all call is very random. I do have a log full log with sip trace and there is nothing wrong, codecs are negotiated OK, actually FS sends the 200OK and a few ms latter FS it sends the bye, see SIP flow. The call flow: incoming --> play audio message --> dial group --> member answer (Bye, in log the  error no write codec) There is no transcoding, the incoming call negotiate G711A as well. I'm trying to figure out what could be causing this, but it looks there is some lock in getting the codec ready, or something is freezing FS. In the code, the only place where we can get this message is at switch_core_media.c :     if (!(session->write_codec && switch_core_codec_ready(session->write_codec)) && !pass_cng) {         switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "%s has no write codec.\n", switch_channel_get_name(session->channel));         switch_channel_hangup(session->channel, SWITCH_CAUSE_INCOMPATIBLE_DESTINATION);         return SWITCH_STATUS_FALSE;     } In sofia profile i've set: So, i guess the only place that is triggering this message is switch_core_codec_ready: static inline switch_bool_t switch_core_codec_ready(switch_codec_t *codec) {     return (codec && (codec->flags & SWITCH_CODEC_FLAG_READY) && codec->mutex && codec->codec_interface && codec->implementation) ? SWITCH_TRUE : SWITCH_FALSE; } Any hint to figure out the issue? On 25/10/2019 14:43, António Silva wrote: > > Hi Mike, > > Thanks for the reply. > > yes, the negation codec is done correctly at least from the output > messages in the logs. > > > I found the source of the problem, for some i have multiple FS pids, > restarting solve my problem., now there is only one. No more drop calls. > > The "defunct" happen because the g729 license-server wasn't fully > started, this is a known issue, but the two main pid's no idea why. > > Just in case i also set in switch.xml the parameter > threaded-system-exec"  to "true". > > > Just for info, the output for "ps aux | grep freeswit" when the > problem happen: > > 0  0.0  0.0      0     0 ?        Z<   Oct07   0:00  \_ [freeswitch] > > > root     11741  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11755  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11756  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11773  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11803  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11804  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     11838  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     12027  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     12194  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     12195  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     12232  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     13966  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14562  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14563  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14564  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14586  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14590  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14630  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14652  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14653  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14670  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14801  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     14802  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16401  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16792  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16899  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16900  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16909  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16910  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16930  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     16931  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17092  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17093  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17128  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17301  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17318  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17319  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17320  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17321  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17341  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17342  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17375  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17376  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17386  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17387  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17407  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17461  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     17462  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     18870  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     18879  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     18905  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     18906  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19084  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19085  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19332  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19333  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19334  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19351  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     19352  0.0  0.0      0     0 ?        Z<   Oct07 0:00  \_ > [freeswitch] > > root     24268  0.0  2.2 1259792 739152 ?      SN   Oct10 0:00  \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > root     24855  0.0  4.2 1920068 1397272 ?     SN   Oct23 0:00  \_ > /usr/bin/freeswitch -ncwait -nonat -scripts /opt/commsmundi/scripts/fs > > > On 24/10/2019 20:35, Mike Jerris wrote: >> Take a look at log with sip trace enabled but this usually happens >> when we can’t negotiate codecs. >> >>> On Oct 24, 2019, at 10:19 AM, António Silva via FreeSWITCH-users >>> >> > wrote: >>> >>> >>> *From: *António Silva >> > >>> *Subject: **call drop because of no write codec, >>> incompatible_destination* >>> *Date: *October 24, 2019 at 9:40:00 AM MDT >>> *To: *FreeSWITCH Users Help >> > >>> >>> >>> Hi, >>> >>> I'm seeing multiple calls drop with incompatible_destination, i try >>> to call again the same destination and it works fine. it gets me >>> crazy... >>> >>> In logs i see the error, "has no write codec",  in the capture i see >>> the negotiation both sides have codecs, they are correctly configure >>> in directory. >>> >>> what could be causing this? i'm using version 1.8.6 >>> >>> >>> Part of the LOG where the error happens: >>> >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:37.694352 >>> [ALERT] switch_core_state_machine.c:701 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 session thread sleep >>> state: CS_CONSUME_MEDIA! >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.794408 >>> [ALERT] switch_core_session.c:1130 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.794408 >>> [ALERT] switch_core_session.c:1130 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.794408 >>> [ALERT] switch_core_state_machine.c:705 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 session thread wake >>> state: CS_CONSUME_MEDIA! >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.814367 >>> [ALERT] switch_core_media.c:486 Looking for zrtp-hash >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.814367 >>> [ALERT] switch_core_media.c:439 Deciding whether to pass zrtp-hash >>> between legs >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.814367 >>> [ALERT] switch_core_media.c:441 CF_ZRTP_PASSTHRU_REQ not set, so not >>> propagating zrtp-hash >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.834519 >>> [DEBUG] sofia.c:7325 Channel >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 entering state >>> [completing][200] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.834519 >>> [DEBUG] sofia.c:7335 Remote SDP: >>> f39e569f-5184-481a-9f67-6185367e2829 v=0 >>> f39e569f-5184-481a-9f67-6185367e2829 o=11 5008 38 IN IP4 192.168.100.11 >>> f39e569f-5184-481a-9f67-6185367e2829 s=Mapping >>> f39e569f-5184-481a-9f67-6185367e2829 c=IN IP4 192.168.100.11 >>> f39e569f-5184-481a-9f67-6185367e2829 t=0 0 >>> f39e569f-5184-481a-9f67-6185367e2829 m=audio 5008 RTP/AVP 0 101 >>> f39e569f-5184-481a-9f67-6185367e2829 a=rtpmap:0 PCMU/8000 >>> f39e569f-5184-481a-9f67-6185367e2829 a=rtpmap:101 telephone-event/8000 >>> f39e569f-5184-481a-9f67-6185367e2829 a=fmtp:101 0-16 >>> f39e569f-5184-481a-9f67-6185367e2829 a=ptime:20 >>> f39e569f-5184-481a-9f67-6185367e2829 >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.834519 >>> [NOTICE] sofia.c:7338 Pre-Answer >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112! >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.934371 >>> [ALERT] switch_core_session.c:2662 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 Send KeyFrame >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.934371 >>> [DEBUG] switch_core_session.c:2825 EXECUTE [depth=1] >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 >>> lua(execute_on_ring_b.lua) >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.934371 >>> [ALERT] switch_core_session.c:2889 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 receive message >>> [APPLICATION_EXEC] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.934371 >>> [ALERT] switch_core_session.c:1047 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.954358 >>> [ALERT] switch_core_session.c:1047 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:41.954358 >>> [ALERT] switch_ivr_originate.c:4136 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 receive message >>> [AUDIO_SYNC] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.034370 >>> [DEBUG] switch_cpp.cpp:1187 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 destroy/unlink >>> session from object >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.034370 >>> [ALERT] switch_core_session.c:2905 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 receive message >>> [APPLICATION_EXEC_COMPLETE] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.034370 >>> [DEBUG] switch_channel.c:3537 >>> (sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112) Callstate Change >>> RINGING -> EARLY >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] sofia.c:7338 sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 >>> receive message [PROGRESS_EVENT] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_media.c:486 Looking for zrtp-hash >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_media.c:439 Deciding whether to pass zrtp-hash >>> between legs >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_media.c:441 CF_ZRTP_PASSTHRU_REQ not set, so not >>> propagating zrtp-hash >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_ivr_bridge.c:1650 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 receive message >>> [AUDIO_SYNC] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_state_machine.c:701 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 session thread sleep >>> state: CS_CONSUME_MEDIA! >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_ivr_bridge.c:1751 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 receive message [BRIDGE] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_session.c:981 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [DEBUG] switch_ivr_bridge.c:1795 >>> (sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112) State Change >>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.094512 >>> [ALERT] switch_core_session.c:1480 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.114382 >>> [ALERT] switch_core_state_machine.c:705 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 session thread wake >>> state: CS_CONSUME_MEDIA! >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.114382 >>> [DEBUG] switch_core_state_machine.c:584 >>> (sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112) Running State >>> Change CS_EXCHANGE_MEDIA (Cur 242 Tot 835614) >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.114382 >>> [DEBUG] switch_core_state_machine.c:653 >>> (sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112) State EXCHANGE_MEDIA >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.114382 >>> [ALERT] switch_core_state_machine.c:653 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 Send KeyFrame >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.114382 >>> [DEBUG] mod_sofia.c:657 SOFIA EXCHANGE_MEDIA >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.134357 >>> [ERR] switch_core_media.c:15507 >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 has no write codec. >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.134357 >>> [NOTICE] switch_core_media.c:15508 Hangup >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [CS_EXCHANGE_MEDIA] >>> [INCOMPATIBLE_DESTINATION] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.134357 >>> [ALERT] switch_channel.c:3346 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [KILL] >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.134357 >>> [DEBUG] mod_hash.c:297 Usage for >>> total_account:11 at clinicalluch.daratel.com >>> is now 0 >>> f39e569f-5184-481a-9f67-6185367e2829 2019-10-21 11:50:42.134357 >>> [ALERT] switch_core_session.c:1480 Send signal >>> sofia/1-perfil-fija/sip:11 at 131.179.185.131:5112 [BREAK] >>> >>> >>> >>> -- >>> Saludos / Regards / Cumprimentos >>> António Silva >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >>> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >>> services. >>> Build your next product on our scalable cloud platform. >>> >>> Join our online community to chat in real time >>> https://signalwire.community >>> >>> Professional FreeSWITCH Services >>> sales at freeswitch.com >>> https://freeswitch.com >>> >>> Official FreeSWITCH Sites >>> https://freeswitch.com/oss >>> https://freeswitch.org/confluence >>> https://cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users 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 -- Saludos / Regards / Cumprimentos António Silva -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Screenshot 2019-11-26 at 17.43.51.png Type: image/png Size: 42399 bytes Desc: not available URL: From jprangi at gmail.com Tue Nov 26 01:44:00 2019 From: jprangi at gmail.com (Jai Rangi) Date: Mon, 25 Nov 2019 17:44:00 -0800 Subject: [Freeswitch-users] Delay in Options Packets In-Reply-To: <0DD51286-4E72-4159-AB48-51FC27CDFF76@freeswitch.org> References: <0DD51286-4E72-4159-AB48-51FC27CDFF76@freeswitch.org> Message-ID: That's what I thought initially. There are around 1000 registrations (mostly TCP) (LUA + Mysql) with timeout of 15 minutes. There are lots of BLFs, subscribes and shared lines, and multiple registrations (average of 3.5, so total of 3500 registrations points). We do ping for keepalive every 3 minutes. I guess increasing ping time and registration timeout to 30 minutes will help. Thoughts? -Jai On Mon, Nov 25, 2019 at 4:33 PM Mike Jerris wrote: > Our code to handle options couldn’t be much simpler. If you are seeing > massive delays the only thing I would guess would be that there is some > sort of massive ddos type attack causing significant queuing of inbound > requests or a register storm and your registrations handlers are getting > behind. Take a look at logging and sip traffic around the same time for > clues on what else might be happening on this sip profile at the same time. > > On Nov 25, 2019, at 12:22 PM, Jai Rangi wrote: > > Hello, > > Every now and then we see long delay in response for options. Example in > the trace below free-switch took 44 seconds before it send 200 OK. > > What will be best way to troubleshoot this? > > U 2019/11/25 10:49:27.185325 192.168.1.1:49020 -> 192.168.1.5:5060 > OPTIONS sip:100000 at 192.168.1.5 SIP/2.0. > Via: SIP/2.0/UDP 192.168.2.1:49020;branch=z9hG4bKhjhs8ass877. > Max-Forwards: 70. > To: sip:100000 at 192.168.1.5. > From: sip:1000 at 192.168.2.1:49020;tag=cbzxq0. > Call-ID: hcrtfi at 192.168.2.1. > CSeq: 1 OPTIONS. > Accept: application/sdp. > Content-Length: 0. > . > > U 2019/11/25 10:49:51.819959 192.168.1.5:5060 -> 192.168.1.1:49020 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.2.1:49020 > ;branch=z9hG4bKhjhs8ass877;received=192.168.1.1. > From: ;tag=cbzxq0. > To: ;tag=aUUQj55rcetma. > Call-ID: hcrtfi at 192.168.2.1. > CSeq: 1 OPTIONS. > Contact: . > User-Agent: Monitor. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE. > Supported: path, replaces. > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, > line-seize, call-info, sla, include-session-description, presence.winfo, > message-summary, refer. > Content-Length: 0. > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users 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 gmaruzz at gmail.com Tue Nov 26 20:33:25 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 26 Nov 2019 21:33:25 +0100 Subject: [Freeswitch-users] Delay in Options Packets In-Reply-To: References: <0DD51286-4E72-4159-AB48-51FC27CDFF76@freeswitch.org> Message-ID: On Tue, Nov 26, 2019 at 7:53 PM Jai Rangi wrote: > That's what I thought initially. There are around 1000 registrations > (mostly TCP) (LUA + Mysql) with timeout of 15 minutes. There are lots of > BLFs, subscribes and shared lines, and multiple registrations (average of > 3.5, so total of 3500 registrations points). We do ping for keepalive every > 3 minutes. I guess increasing ping time and registration timeout to 30 > minutes will help. > Thoughts? > > You may want to uncomment (or add if not there) the second line: in sip_profiles/internal.xml -giovanni > > -Jai > > > > On Mon, Nov 25, 2019 at 4:33 PM Mike Jerris wrote: > >> Our code to handle options couldn’t be much simpler. If you are seeing >> massive delays the only thing I would guess would be that there is some >> sort of massive ddos type attack causing significant queuing of inbound >> requests or a register storm and your registrations handlers are getting >> behind. Take a look at logging and sip traffic around the same time for >> clues on what else might be happening on this sip profile at the same time. >> >> On Nov 25, 2019, at 12:22 PM, Jai Rangi wrote: >> >> Hello, >> >> Every now and then we see long delay in response for options. Example in >> the trace below free-switch took 44 seconds before it send 200 OK. >> >> What will be best way to troubleshoot this? >> >> U 2019/11/25 10:49:27.185325 192.168.1.1:49020 -> 192.168.1.5:5060 >> OPTIONS sip:100000 at 192.168.1.5 SIP/2.0. >> Via: SIP/2.0/UDP 192.168.2.1:49020;branch=z9hG4bKhjhs8ass877. >> Max-Forwards: 70. >> To: sip:100000 at 192.168.1.5. >> From: sip:1000 at 192.168.2.1:49020;tag=cbzxq0. >> Call-ID: hcrtfi at 192.168.2.1. >> CSeq: 1 OPTIONS. >> Accept: application/sdp. >> Content-Length: 0. >> . >> >> U 2019/11/25 10:49:51.819959 192.168.1.5:5060 -> 192.168.1.1:49020 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 192.168.2.1:49020 >> ;branch=z9hG4bKhjhs8ass877;received=192.168.1.1. >> From: ;tag=cbzxq0. >> To: ;tag=aUUQj55rcetma. >> Call-ID: hcrtfi at 192.168.2.1. >> CSeq: 1 OPTIONS. >> Contact: . >> User-Agent: Monitor. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. >> Supported: path, replaces. >> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, >> line-seize, call-info, sla, include-session-description, presence.winfo, >> message-summary, refer. >> Content-Length: 0. >> >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From thomas.peterseil at mine-project.eu Wed Nov 27 16:54:11 2019 From: thomas.peterseil at mine-project.eu (thomas peterseil) Date: Wed, 27 Nov 2019 17:54:11 +0100 Subject: [Freeswitch-users] bash script with variables Message-ID: hello freeswitch-list, i would like to start from the dialplan a bash script and the freeswitch is handing over one variable to the script. in the cli i see the correct start of the script: [NOTICE] mod_dptools.c:2120 Executing command: /root/scripts/test1.sh 3333 the script is very simple: #!/bin/bash number=$1 mutt -s "Call $number" thomas.peterseil at mine-project.eu < /root/mailtexte/registrierung.txt echo "$number" > test1.txt i get the email with the variable in the subject, that works fine, but i can´t see the 3333 in the test1.txt file. when i start the script from the command line with ./test1.sh 3333 all is working fine. can somebody give me a hint why it doesn´t work from the dialplan. thanks a lot and have a nice day! best regards, thomas From mike at freeswitch.org Wed Nov 27 20:30:56 2019 From: mike at freeswitch.org (Mike Jerris) Date: Wed, 27 Nov 2019 12:30:56 -0800 Subject: [Freeswitch-users] bash script with variables In-Reply-To: References: Message-ID: <95CD35A0-EC6A-4F76-B28E-4C3D19BA073B@freeswitch.org> Do you not see the file at all? Might be a permissions or working directory issue. > On Nov 27, 2019, at 8:54 AM, thomas peterseil wrote: > > hello freeswitch-list, > i would like to start from the dialplan a bash script and the > freeswitch is handing over one variable to the script. in the cli i > see the correct start of the script: > > [NOTICE] mod_dptools.c:2120 Executing command: /root/scripts/test1.sh 3333 > > the script is very simple: > > #!/bin/bash > number=$1 > > mutt -s "Call $number" thomas.peterseil at mine-project.eu < > /root/mailtexte/registrierung.txt > echo "$number" > test1.txt > > i get the email with the variable in the subject, that works fine, but > i can´t see the 3333 in the test1.txt file. > when i start the script from the command line with ./test1.sh 3333 all > is working fine. can somebody give me a hint why it doesn´t work from > the dialplan. From ch.chhatra at gmail.com Thu Nov 28 06:28:38 2019 From: ch.chhatra at gmail.com (Chhorm Chhatra) Date: Thu, 28 Nov 2019 13:28:38 +0700 Subject: [Freeswitch-users] Freeswitch 1.8 mod_callcenter get CALL_REJECTED Originate Error Message-ID: Hi, Currently, our Freeswitch 1.8 production server, frequently encounters CALL_REJECTED when mod_callcenter tries to originate the call to the agent. *Is there any clue you can suggest to debug this issue?* I am sure that the callee does not reject this call because mod_callcenter > can't even make the call to the agent (Callee). The call would traverse like this: *DID Call via GSM Gateway > Freeswitch > (IVR Call > Callcenter-Queue > Agent)* Any help would be really appreciated. Best regards, Chhorm Chhatra -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: incident-xx-xx-2019.log Type: text/x-log Size: 2753 bytes Desc: not available URL: From dragos at freeswitch.org Thu Nov 28 12:55:14 2019 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 28 Nov 2019 12:55:14 +0000 Subject: [Freeswitch-users] Local stream In-Reply-To: <16367039-4530-4A69-989B-E8D22036A9FF@groupechaka.com> References: <16367039-4530-4A69-989B-E8D22036A9FF@groupechaka.com> Message-ID: That's not an error, is merely a piece of information that FS is about to resample. You're supposed to use the 8000 hz section for 8000 hz moh. If you dropped the file in the 16000 directory it won't find it with 8000 hz calls. On Thu, Nov 7, 2019 at 6:08 PM Abdou khadre NDOYE wrote: > Hi, > > I have some trouble using mod_local_stream, > > When I place a file in the right path configured in local_stream.conf.xml > file doesn’t played. > > Sometime I have this error when I use rate 8000 : > > 2019-11-07 10:53:14.620285 [DEBUG] switch_core_media.c:8494 Audio params > are unchanged for sofia/internal/338600485 at 172.16.10.5. > 2019-11-07 10:53:14.660286 [DEBUG] switch_ivr.c:625 sofia/internal/ > 315 at 172.16.10.50:5060 Command Execute playback(local_stream://moh) > EXECUTE sofia/internal/315 at 172.16.10.50:5060 playback(local_stream://moh) > 2019-11-07 10:53:14.660286 [DEBUG] switch_core_file.c:389 File moh sample > rate 8000 doesn't match requested rate 16000 > 2019-11-07 10:53:14.660286 [DEBUG] switch_ivr_play_say.c:1497 Codec > Activated L16 at 16000hz 1 channels 20ms > 2019-11-07 10:53:14.680286 [NOTICE] switch_core_media.c:15605 Deactivating > write resampler > > My config ; > > > > > > > > > > I need help. > > Thank > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From dragos at freeswitch.org Thu Nov 28 13:14:27 2019 From: dragos at freeswitch.org (Dragos Oancea) Date: Thu, 28 Nov 2019 13:14:27 +0000 Subject: [Freeswitch-users] bash script with variables In-Reply-To: <95CD35A0-EC6A-4F76-B28E-4C3D19BA073B@freeswitch.org> References: <95CD35A0-EC6A-4F76-B28E-4C3D19BA073B@freeswitch.org> Message-ID: Maybe it has something to do with mutt config and maybe mutt is even clearing an env variable with the same name when it's getting executed as user 'freeswitch' from the dialplan. When you execute that script from command line you do it as a regular user, try to become the freeswitch user (su) and see if it still works. On Wed, Nov 27, 2019 at 8:31 PM Mike Jerris wrote: > Do you not see the file at all? Might be a permissions or working > directory issue. > > > On Nov 27, 2019, at 8:54 AM, thomas peterseil < > thomas.peterseil at mine-project.eu> wrote: > > > > hello freeswitch-list, > > i would like to start from the dialplan a bash script and the > > freeswitch is handing over one variable to the script. in the cli i > > see the correct start of the script: > > > > [NOTICE] mod_dptools.c:2120 Executing command: /root/scripts/test1.sh > 3333 > > > > the script is very simple: > > > > #!/bin/bash > > number=$1 > > > > mutt -s "Call $number" thomas.peterseil at mine-project.eu < > > /root/mailtexte/registrierung.txt > > echo "$number" > test1.txt > > > > i get the email with the variable in the subject, that works fine, but > > i can´t see the 3333 in the test1.txt file. > > when i start the script from the command line with ./test1.sh 3333 all > > is working fine. can somebody give me a hint why it doesn´t work from > > the dialplan. > > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Thu Nov 28 15:23:51 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Thu, 28 Nov 2019 20:23:51 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] Message-ID: Hi users, I am using mod_avmd for quite a while, but i now want to use amd like its in asterisk. I found a related module here https://github.com/seanbright/mod_amd I am not able to successfully generate events through that, do someone has a working dialplan example along with parameters. That would be really helpful. P.S: i have tested on freeswitch 1.10, any special version for this. Regards Abbasi -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Thu Nov 28 22:59:51 2019 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 28 Nov 2019 19:59:51 -0300 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: >From GitHub: Currently, in limited testing, we are able to get satisfactory results in determining what is a human and what is a machine, but *there is much more to do*: - Emit events when a decision is made (Not sure, Machine, or Human). <== This is a TODO item - Make sure that we are unlocking and cleaning up where necessary. The last commit was 4 years ago. It looks like this is an abandoned project. Guillermo On Thu, Nov 28, 2019 at 1:01 PM Bilal Abbasi wrote: > Hi users, > I am using mod_avmd for quite a while, but i now want to use amd like its > in asterisk. > I found a related module here > https://github.com/seanbright/mod_amd > > I am not able to successfully generate events through that, do someone has > a working dialplan example along with parameters. That would be > really helpful. > > P.S: i have tested on freeswitch 1.10, any special version for this. > > Regards > Abbasi > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From ynasida at gmail.com Fri Nov 29 09:05:45 2019 From: ynasida at gmail.com (Yuriy Nasida) Date: Fri, 29 Nov 2019 12:05:45 +0300 Subject: [Freeswitch-users] segfault because of require("ESL") in lua Message-ID: Hi guys, FS runs my lua script during the call. At the same time this script should run some fs_cli commands at different fs node. I want to invoke ESL from lua to do this. Please advice is it good idea ? Currently FS got segfault because of require("ESL") in lua. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bilaln018 at gmail.com Fri Nov 29 09:51:37 2019 From: bilaln018 at gmail.com (Bilal Abbasi) Date: Fri, 29 Nov 2019 14:51:37 +0500 Subject: [Freeswitch-users] [Using amd in freeswitch] In-Reply-To: References: Message-ID: Hi, Thanks for your reply, is there any way i can achieve this via wait_for_silence, i am just wondering if some one has done this in freeswitch or even this is something possible. Regards Abbasi On Fri, 29 Nov 2019 at 4:54 AM, Guillermo Ruiz Camauer wrote: > From GitHub: > > Currently, in limited testing, we are able to get satisfactory results in > determining what is a human and what is a machine, but *there is much > more to do*: > > - Emit events when a decision is made (Not sure, Machine, or Human). > <== This is a TODO item > - Make sure that we are unlocking and cleaning up where necessary. > > The last commit was 4 years ago. It looks like this is an abandoned > project. > > Guillermo > > On Thu, Nov 28, 2019 at 1:01 PM Bilal Abbasi wrote: > >> Hi users, >> I am using mod_avmd for quite a while, but i now want to use amd like its >> in asterisk. >> I found a related module here >> https://github.com/seanbright/mod_amd >> >> I am not able to successfully generate events through that, do someone >> has a working dialplan example along with parameters. That would be >> really helpful. >> >> P.S: i have tested on freeswitch 1.10, any special version for this. >> >> Regards >> Abbasi >> > _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN >> services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time >> https://signalwire.community >> >> Professional FreeSWITCH Services >> sales at freeswitch.com >> https://freeswitch.com >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss >> https://freeswitch.org/confluence >> https://cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> https://freeswitch.com > > > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Fri Nov 29 15:46:21 2019 From: covici at ccs.covici.com (John Covici) Date: Fri, 29 Nov 2019 10:46:21 -0500 Subject: [Freeswitch-users] problems with portaudio Message-ID: Hi. I finally was able to upgrade fs to master as of llast night. Its working well, except if I use portaudio to make a call. This all worked find in fs 1.6.20. When I call someone I cannot hear anything until I send it a dtmf (rfc2283) and then things work normally, at least I can hear something. I had a look at the logs, but nothing strange in there after typing the digit. Also, I cannot call a local extension from port audio, even though the extension is registered and can be called from another extension. It immediately goes to voicemail. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From covici at ccs.covici.com Fri Nov 29 17:45:59 2019 From: covici at ccs.covici.com (John Covici) Date: Fri, 29 Nov 2019 12:45:59 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: Message-ID: Some more information -- even after pressing a digit and getting audio, it hangs up after about 30 seconds. On Fri, 29 Nov 2019 10:46:21 -0500, John Covici wrote: > > Hi. I finally was able to upgrade fs to master as of llast night. > Its working well, except if I use portaudio to make a call. This all > worked find in fs 1.6.20. > > When I call someone I cannot hear anything until I send it a dtmf > (rfc2283) and then things work normally, at least I can hear > something. I had a look at the logs, but nothing strange in there > after typing the digit. > > Also, I cannot call a local extension from port audio, even though the > extension is registered and can be called from another extension. It > immediately goes to voicemail. > > Thanks in advance for any suggestions. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From david.villasmil.work at gmail.com Fri Nov 29 18:06:12 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 29 Nov 2019 18:06:12 +0000 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: Message-ID: Do you have any trace? On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > Some more information -- even after pressing a digit and getting > audio, it hangs up after about 30 seconds. > > On Fri, 29 Nov 2019 10:46:21 -0500, > John Covici wrote: > > > > Hi. I finally was able to upgrade fs to master as of llast night. > > Its working well, except if I use portaudio to make a call. This all > > worked find in fs 1.6.20. > > > > When I call someone I cannot hear anything until I send it a dtmf > > (rfc2283) and then things work normally, at least I can hear > > something. I had a look at the logs, but nothing strange in there > > after typing the digit. > > > > Also, I cannot call a local extension from port audio, even though the > > extension is registered and can be called from another extension. It > > immediately goes to voicemail. > > > > Thanks in advance for any suggestions. > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici wb2una > > covici at ccs.covici.com > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales at freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > covici at ccs.covici.com > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Fri Nov 29 19:28:21 2019 From: covici at ccs.covici.com (John Covici) Date: Fri, 29 Nov 2019 14:28:21 -0500 Subject: [Freeswitch-users] problems with portaudio In-Reply-To: References: Message-ID: I have the log of the call which looks normal. My guess is that rtp is not properly being sent out, for some reason. The hangup cause is always normal_clearing. On Fri, 29 Nov 2019 13:06:12 -0500, David Villasmil wrote: > > [1 ] > [1.1 ] > Do you have any trace? > > On Fri, 29 Nov 2019 at 18:05, John Covici wrote: > > > Some more information -- even after pressing a digit and getting > > audio, it hangs up after about 30 seconds. > > > > On Fri, 29 Nov 2019 10:46:21 -0500, > > John Covici wrote: > > > > > > Hi. I finally was able to upgrade fs to master as of llast night. > > > Its working well, except if I use portaudio to make a call. This all > > > worked find in fs 1.6.20. > > > > > > When I call someone I cannot hear anything until I send it a dtmf > > > (rfc2283) and then things work normally, at least I can hear > > > something. I had a look at the logs, but nothing strange in there > > > after typing the digit. > > > > > > Also, I cannot call a local extension from port audio, even though the > > > extension is registered and can be called from another extension. It > > > immediately goes to voicemail. > > > > > > Thanks in advance for any suggestions. > > > > > > -- > > > Your life is like a penny. You're going to lose it. The question is: > > > How do > > > you spend it? > > > > > > John Covici wb2una > > > covici at ccs.covici.com > > > > > > _________________________________________________________________________ > > > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > > services. > > > Build your next product on our scalable cloud platform. > > > > > > Join our online community to chat in real time > > https://signalwire.community > > > > > > Professional FreeSWITCH Services > > > sales at freeswitch.com > > > https://freeswitch.com > > > > > > Official FreeSWITCH Sites > > > https://freeswitch.com/oss > > > https://freeswitch.org/confluence > > > https://cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > https://freeswitch.com > > > > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici wb2una > > covici at ccs.covici.com > > > > _________________________________________________________________________ > > > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > > services. > > Build your next product on our scalable cloud platform. > > > > Join our online community to chat in real time > > https://signalwire.community > > > > Professional FreeSWITCH Services > > sales at freeswitch.com > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > https://freeswitch.com > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > [1.2 ] > [2 ] > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici at ccs.covici.com From tafali at gmail.com Fri Nov 29 13:47:43 2019 From: tafali at gmail.com (Mustafa Ali Kahraman) Date: Fri, 29 Nov 2019 16:47:43 +0300 Subject: [Freeswitch-users] call transfer Message-ID: hi, i want to transfer calls from one provider to another one. i've done like this. *Questions*: 1- is this the right way? 2- when i do like this, do RTP packages go over my server? 3- what should i do, just to transfer signalling, not to transfer media over my server? thanks. -- -tafali- www.tafali.net -------------- next part -------------- An HTML attachment was scrubbed... URL: From ciprian.dosoftei at gmail.com Fri Nov 29 22:48:49 2019 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Fri, 29 Nov 2019 17:48:49 -0500 Subject: [Freeswitch-users] call transfer In-Reply-To: References: Message-ID: > > 1- is this the right way? > It really depends on your circumstances and the use case, so it's difficult to tell. Also, the network address criteria may not be as stable as expected, perhaps another variable is better suited for that condition. > 2- when i do like this, do RTP packages go over my server? > In most circumstances, the RTP will flow through your server indeed. > 3- what should i do, just to transfer signalling, not to transfer media > over my server? > Check bypass_media and/or bypass_media_after_bridge: https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#VariablesMasterList-bypass_media -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Nov 30 00:14:37 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 30 Nov 2019 00:14:37 +0000 Subject: [Freeswitch-users] call transfer In-Reply-To: References: Message-ID: Iif you're just forwarding to another destination, you'd better let the destination answer the call, instead of you answering _after_ bridging it: <-- As Ciprian pointed out, using the IP is not always a good idea. Are you also using ACL? <--- shouldn't be here, you're answering the call after you've transferred it. You're also not using a gateway, but an IP address, instead create a gateway and use it like so: About the media, according to Brian: bypass_media=false > Does nothing. > proxy_media=false > Does nothing. But that was in regards to t.38, so... Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Fri, Nov 29, 2019 at 11:03 PM Ciprian Dosoftei < ciprian.dosoftei at gmail.com> wrote: > 1- is this the right way? >> > > It really depends on your circumstances and the use case, so it's > difficult to tell. Also, the network address criteria may not be as stable > as expected, perhaps another variable is better suited for that condition. > > >> 2- when i do like this, do RTP packages go over my server? >> > > In most circumstances, the RTP will flow through your server indeed. > > >> 3- what should i do, just to transfer signalling, not to transfer media >> over my server? >> > > Check bypass_media and/or bypass_media_after_bridge: > https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#VariablesMasterList-bypass_media > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales at freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -------------- next part -------------- An HTML attachment was scrubbed... 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