From leon.ramos at tiendaip.mx Sat Nov 1 00:41:18 2014 From: leon.ramos at tiendaip.mx (leon.ramos at tiendaip.mx) Date: Fri, 31 Oct 2014 16:41:18 -0500 Subject: [Freeswitch-users] Turbina-base Freeswitch ISO In-Reply-To: <5453ED5B.4090509@quentustech.com> References: <20141031143343.14772k59frf8a0bb@webmail.tiendaip.mx> <5453ED5B.4090509@quentustech.com> Message-ID: <20141031164118.3302779fxan2bfni@webmail.tiendaip.mx> William, You are absolutely right, my bad. It is based on fs 1.4.12 sorry about that. Thanks William King escribi?: > I'm very curious how you have freeswitch 1.4.20 when that hasn't been > tagged in tree. If you mean 1.4.2 then that's really old and you should > update. > > -William > > On 10/31/2014 12:33 PM, leon.ramos at tiendaip.mx wrote: >> Hi, >> >> We recently released a new ISO based on CentOS 6.5, we repackaged a >> freeswitch 1.4.20 and added libpri, dahdi and libopenr2 (1.2). English >> and spanish 8000Hz are included. >> >> Feel free to download and give any feedback for improvements! >> >> https://sourceforge.net/projects/turbina/files/base-system/ >> >> Nice friday! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ssinyagin at gmail.com Sat Nov 1 05:21:02 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sat, 1 Nov 2014 03:21:02 +0100 Subject: [Freeswitch-users] Turbina-base Freeswitch ISO In-Reply-To: <20141031164118.3302779fxan2bfni@webmail.tiendaip.mx> References: <20141031143343.14772k59frf8a0bb@webmail.tiendaip.mx> <5453ED5B.4090509@quentustech.com> <20141031164118.3302779fxan2bfni@webmail.tiendaip.mx> Message-ID: Just tell that your time machine will be accidentally set to a wrong date. But what's the point of using an ISO if you can install from packages or sources? Besides, it limits you to the hardware where you boot from ISO, while in many VPS environments you can't. On Nov 1, 2014 1:21 AM, wrote: > William, > > You are absolutely right, my bad. It is based on fs 1.4.12 sorry about > that. > > Thanks > > > > William King escribi?: > > > I'm very curious how you have freeswitch 1.4.20 when that hasn't been > > tagged in tree. If you mean 1.4.2 then that's really old and you should > > update. > > > > -William > > > > On 10/31/2014 12:33 PM, leon.ramos at tiendaip.mx wrote: > >> Hi, > >> > >> We recently released a new ISO based on CentOS 6.5, we repackaged a > >> freeswitch 1.4.20 and added libpri, dahdi and libopenr2 (1.2). English > >> and spanish 8000Hz are included. > >> > >> Feel free to download and give any feedback for improvements! > >> > >> https://sourceforge.net/projects/turbina/files/base-system/ > >> > >> Nice friday! > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141101/46875044/attachment.html From mvar78 at gmail.com Sat Nov 1 16:30:58 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Sat, 1 Nov 2014 14:30:58 +0100 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer Message-ID: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> Hi All, is possible to create a scenario in which I have a postpaid customer in which the credit can go below zero but no more than a maximum value allowed? The wiki talks about the feature of having a postpaid customer, but there are no example regarding this feature. But on prepaid there are many and many example, thank for this, great! BTW, currently I can create a prepaid customer and as soon as credit reach 0 the call is dropped correctly, but how can I implement a postpaid customer? Thank you so much Max From krice at freeswitch.org Sat Nov 1 17:10:01 2014 From: krice at freeswitch.org (Ken Rice) Date: Sat, 1 Nov 2014 09:10:01 -0500 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> Message-ID: <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> you just allow a negative balance for a post paid customer Ken Sent from my iPad > On Nov 1, 2014, at 8:30, Massimo Varriale wrote: > > Hi All, > is possible to create a scenario in which I have a postpaid customer in which the credit can go below zero but no more than a maximum value allowed? > The wiki talks about the feature of having a postpaid customer, but there are no example regarding this feature. But on prepaid there are many and many example, thank for this, great! > > BTW, currently I can create a prepaid customer and as soon as credit reach 0 the call is dropped correctly, but how can I implement a postpaid customer? > > Thank you so much > Max > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mvar78 at gmail.com Sat Nov 1 17:11:28 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Sat, 1 Nov 2014 15:11:28 +0100 Subject: [Freeswitch-users] [Lua Script] Conditional Remapping of Hangup Cause Message-ID: Hi, I'm creating a Lua script for managing different call scenario. At the moment I'm stuck with a problem regarding the interoperability between Freeswitch and some potential customer. The problem regards the hangup cause that Freeswitch is sending to their softswitch, to allow a their automatic rerouting based on some Hangup Cause. I got a request to send different Hangup Cause, so the Customer switch can automatically reroute call to another carrier. Here is the explaination The scenario is: All the endpoints are using a Public IP and no NAT is involved in these steps. CUSTOMER ------------------> MY_FREESWITCH -----------------> REMOTE_CARRIER What happen now: The customer can send calls. On My freeswitch I will route calls based on Dialled Prefix to a Remote Carrier. In my Lua script there is a conditional IF to check for remote disconnect cause, however it seems that the Release Cause is already sent (CALL_REJECTED) when I'm checking with my IF and so, even if the cause has been already sent it's sent again but what "win" is the first hangup sent. What should be: If the remote carrier will send me a CALL_REJECTED (isdn 21) I need to send to the remote customer another Hangup Cause for example: NORMAL_CIRCUIT_CONGESTION (isdn 34). How to implement that? I can create this remapping using a Diaplan XML configuration, but as I'm using a Lua script to manage the call, I found that is a Lua task to deal with the Hangup Cause, the dialplan rules are ignored. The "telephony" part of the Lua script is working, what I'm trying to change is the release cause to make my customer happy as with a different release cause he can route traffic correctly and I will not loose my reputation :) Below there is my Lua script (just the routing part) Thank you so much! Cheers Max > function string.starts(String,Start) > return string.sub(String,1,string.len(Start))==Start > end > > session:execute("set","hangup_after_bridge=true") > called_number = session:getVariable("destination_number") > > > while session:answered()==false and session:ready()==true do > > if string.starts(called_number, "393") then > -- ITALY MOBILE > route_dial_string = "sofia/gateway/REMOTE_CARRIER/" ..called_number > else > -- OTHER DESTINATIONS > route_dial_string = "sofia/gateway/OTHER_CARRIER/" ..called_number > end > > session:execute("bridge",""..route_dial_string.."") > > > -- ATTEMPT TO REMAP RELEASE CAUSE > if ( session:getVariable("last_bridge_hangup_cause") == "CALL_REJECTED" ) then > session:hangup("NORMAL_CIRCUIT_CONGESTION"); > end > > > if session:ready()==false then > session:hangup(); > > > end > > > session:hangup(); > freeswitch.consoleLog("info", "That's all folks!"); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141101/8daaba26/attachment.html From mvar78 at gmail.com Sat Nov 1 17:45:45 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Sat, 1 Nov 2014 15:45:45 +0100 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> Message-ID: <4C34B3B1-2B86-41FB-B73A-644C20A277BE@gmail.com> Hi Ken! Is there an option for this? Because as fas as I know I'm setting into the Dialplan nibble_account nibble_rate And into the mod nibblebill conf I'm just setting up columns and that's it. The default behaviour is to drop call as the credit reach 0, I didn't found any other option for postpaid. Thanks Max Il giorno 01/nov/2014, alle ore 15:10, Ken Rice ha scritto: > you just allow a negative balance for a post paid customer > > Ken > Sent from my iPad > >> On Nov 1, 2014, at 8:30, Massimo Varriale wrote: >> >> Hi All, >> is possible to create a scenario in which I have a postpaid customer in which the credit can go below zero but no more than a maximum value allowed? >> The wiki talks about the feature of having a postpaid customer, but there are no example regarding this feature. But on prepaid there are many and many example, thank for this, great! >> >> BTW, currently I can create a prepaid customer and as soon as credit reach 0 the call is dropped correctly, but how can I implement a postpaid customer? >> >> Thank you so much >> Max >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sos at sokhapkin.dyndns.org Sat Nov 1 19:31:01 2014 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 01 Nov 2014 12:31:01 -0400 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <4C34B3B1-2B86-41FB-B73A-644C20A277BE@gmail.com> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> <4C34B3B1-2B86-41FB-B73A-644C20A277BE@gmail.com> Message-ID: <3297783.TD4EZ6Yq2I@sos> Set variable nobal_amt to negative value. On Saturday 01 November 2014 15:45:45 Massimo Varriale wrote: > Hi Ken! > Is there an option for this? > > Because as fas as I know I'm setting into the Dialplan > nibble_account > nibble_rate > > And into the mod nibblebill conf I'm just setting up columns and that's it. > > The default behaviour is to drop call as the credit reach 0, I didn't found > any other option for postpaid. > > Thanks > Max > > Il giorno 01/nov/2014, alle ore 15:10, Ken Rice ha scritto: > > you just allow a negative balance for a post paid customer > > > > Ken > > Sent from my iPad > > > >> On Nov 1, 2014, at 8:30, Massimo Varriale wrote: > >> > >> Hi All, > >> is possible to create a scenario in which I have a postpaid customer in > >> which the credit can go below zero but no more than a maximum value > >> allowed? The wiki talks about the feature of having a postpaid customer, > >> but there are no example regarding this feature. But on prepaid there > >> are many and many example, thank for this, great! > >> > >> BTW, currently I can create a prepaid customer and as soon as credit > >> reach 0 the call is dropped correctly, but how can I implement a > >> postpaid customer? > >> > >> Thank you so much > >> Max > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ben at langfeld.co.uk Sun Nov 2 00:28:49 2014 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sat, 1 Nov 2014 19:28:49 -0200 Subject: [Freeswitch-users] To decypt webrtc audio In-Reply-To: References: Message-ID: Ah, this is not what I understood from the original question. My apologies for the detour. On 31 October 2014 12:38, Varghese Paul wrote: > Hi Ben, > > I am looking for a tool like wireshark to decrypt the SRTP packets and > listen to the audio. RTPengine as you said we can bridge two channels and > we can't use it as a tool for decrypting SRTP traffic, > > Regards > > On Fri, Oct 31, 2014 at 7:25 PM, Carlos Ruiz D?az < > carlos.ruizdiaz at gmail.com> wrote: > >> I understand that this is not what he is intending. Why would he need yet >> another rtp bridge if FS already does exactly that. >> On Oct 31, 2014 7:36 AM, "Ben Langfeld" wrote: >> >>> The way I use rtpengine, it rewrites SDP between the browser and >>> Asterisk 1.4, terminating DTLS and ICE. Between the browser and rtpengine >>> is RTP/SAVPF, while between rtpengine and Asterisk is RTP/AVP. rtpengine >>> generates its keys in what I assume to be the same way the browsers do. >>> >>> On 31 October 2014 10:51, Anthony Minessale >> > wrote: >>> >>>> The tool from pjsip mentioned above requires the keys for the srtp >>>> decryption. Does the one you mention do it differently? >>>> >>>> >>>> On Friday, October 31, 2014, Ben Langfeld wrote: >>>> >>>>> Anthony, I think I'm not understanding your concern. Could you explain >>>>> what you mean by logging the keys? >>>>> >>>>> On 30 October 2014 21:01, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> The bigger question is how to log the keys without compromising the >>>>>> point of why its all encrypted. >>>>>> It would take some careful consideration. >>>>>> >>>>>> >>>>>> On Thu, Oct 30, 2014 at 12:32 PM, Ben Langfeld >>>>>> wrote: >>>>>> >>>>>>> rtpengine does indeed allow decryption of DTLS encrypted RTP. I use >>>>>>> it in front of Asterisk 1.4 for exactly that (and ICE resolution). >>>>>>> >>>>>>> On 30 October 2014 14:55, Carlos Ruiz D?az < >>>>>>> carlos.ruizdiaz at gmail.com> wrote: >>>>>>> >>>>>>>> rtpengine doesn't provide any mean to do any kind of decryption. >>>>>>>> The keys, however, are easily accessible if you are willing to modify the >>>>>>>> original software to create such functionality. >>>>>>>> >>>>>>>> Regards, >>>>>>>> Carlos >>>>>>>> >>>>>>>> On Thu, Oct 30, 2014 at 10:47 AM, Anthony Minessale < >>>>>>>> anthony.minessale at gmail.com> wrote: >>>>>>>> >>>>>>>>> This is a philosophical debate as to weather or not making the key >>>>>>>>> available somewhere it can be obtained from nullifies the intended security. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Thu, Oct 30, 2014 at 5:33 AM, Ben Langfeld >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>>> Look into rtpengine and perhaps Kamailio to control it. >>>>>>>>>> >>>>>>>>>> On 30 October 2014 04:41, Varghese Paul >>>>>>>>> > wrote: >>>>>>>>>> >>>>>>>>>>> Hi all, >>>>>>>>>>> >>>>>>>>>>> We are using freeswitch for handling webrtc calls and the >>>>>>>>>>> negotiation in DTLS mode. >>>>>>>>>>> >>>>>>>>>>> We are looking for a tool to decrypt the traffic between the >>>>>>>>>>> webrtc client and freeswitch server. I know there is already a tool >>>>>>>>>>> pcaputils from PJSIP for decrypting the SRTP traffic. Since we are using in >>>>>>>>>>> DTLS mode we can't use this. >>>>>>>>>>> >>>>>>>>>>> Can any one suggest any tool/method to decrypt the webrtc audio? >>>>>>>>>>> >>>>>>>>>>> Regards >>>>>>>>>>> >>>>>>>>>>> Varghese Paul >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>>>>>> >>>>>>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>>>>>> * >>>>>>>>> >>>>>>>>> ClueCon Weekly Development Call >>>>>>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Carlos >>>>>>>> http://caruizdiaz.com >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>>> >>>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>>> http://twitter.com/FreeSWITCH >>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>>> * >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>> >>>> -- >>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>> >>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>> http://twitter.com/FreeSWITCH >>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141101/c69d8bcd/attachment-0001.html From stesasso at gmail.com Sun Nov 2 16:15:00 2014 From: stesasso at gmail.com (Stefano Sasso) Date: Sun, 2 Nov 2014 14:15:00 +0100 Subject: [Freeswitch-users] codec2 Message-ID: Hello folks, someone of you is using in real life environment (or did some test with) the codec "CODEC2"? I tried to use it with csipsimple UA, but I think I'm having some transcoding issues - but I don't know if the problem is in csipsimple or in freeswitch. (audio passthrough csipsimple<->csipsimple works fine) Do you know some other UA supporting that codec? bests, stefano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141102/873cd6ab/attachment.html From vbvbrj at gmail.com Sun Nov 2 18:16:13 2014 From: vbvbrj at gmail.com (Mimiko) Date: Sun, 02 Nov 2014 17:16:13 +0200 Subject: [Freeswitch-users] Caller number when listening message. Message-ID: <54564ABD.4020409@gmail.com> Hello. It seams that vm module plays only message number and message datetime when listening to stored voice messages. How to enable vm to spell the caller number also? And disable spelling the seconds value from datetime? Thank you. -- Mimiko desu. From nick.zaitsev at mail.ru Mon Nov 3 10:12:49 2014 From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=) Date: Mon, 03 Nov 2014 10:12:49 +0300 Subject: [Freeswitch-users] =?utf-8?q?SCCP=2Cbuttons_in_phone_configs?= Message-ID: <1414998769.410828978@f125.i.mail.ru> Good day to you,maybe it is silly question,but, I actually can't understand how tel buttons can be set for sccp phones in freeswitch. For example, how can i set ?the redial button for? Cisco IP Phone 7961 -- Nick Zaitsev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/b8e993ce/attachment.html From balazs.sandor at virtual-call-center.eu Mon Nov 3 12:25:56 2014 From: balazs.sandor at virtual-call-center.eu (=?UTF-8?B?QmFsw6F6cyBTw6FuZG9y?=) Date: Mon, 3 Nov 2014 10:25:56 +0100 Subject: [Freeswitch-users] mod_verto unwanted profile shutdown Message-ID: Hi all. I'm testing mod_verto. After a while (5-10 minutes) it always disconnecting. I do nothing with FS (its sleeping) It seems to be normal shutdown 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert specified 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8081 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8082 -- 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, Waiting for 0 threads 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending I attached a detailed log Best regards, Bal?zs S?NDOR Software Developer Virtual Call Center MUNICH | BUDAPEST | WARSAW Phone: +44 (0) 863 801 69 Web: www.virtual-call-center.eu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/21ebf5aa/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: verto.log Type: text/x-log Size: 19216 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/21ebf5aa/attachment-0001.bin From matsumoto at itsherpa.com Mon Nov 3 14:11:10 2014 From: matsumoto at itsherpa.com (=?UTF-8?B?5p2+5pys56WQ5b+X?=) Date: Mon, 3 Nov 2014 20:11:10 +0900 Subject: [Freeswitch-users] about conference play and bgapi Message-ID: Hello I have two issues. I am writing in Perl. While 2 people are talking in a conference room, the one person want to play the sound. In "Caller-Username", can you get useless. I have tried the above but, Member: it will not become a *** not found.. my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; I want to play the above in the background It can not play in the next program. my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav); $ con-> bgapi ($ api_cmd); Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/a8fc31f8/attachment.html From mike at jerris.com Mon Nov 3 16:40:03 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Nov 2014 08:40:03 -0500 Subject: [Freeswitch-users] mod_verto unwanted profile shutdown In-Reply-To: References: Message-ID: <1853D18C-7264-45FF-9B17-29487807EB48@jerris.com> Does it do the same thing if you bind to a real ip instead of 0.0.0.0? > On Nov 3, 2014, at 4:25 AM, Bal?zs S?ndor wrote: > > Hi all. > > I'm testing mod_verto. > After a while (5-10 minutes) it always disconnecting. > I do nothing with FS (its sleeping) > > It seems to be normal shutdown > 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert specified > 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8081 > 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to 0.0.0.0:8082 > -- > 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, Waiting for 0 threads > 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending > > I attached a detailed log > > Best regards, > > Bal?zs S?NDOR > Software Developer > > > > Virtual Call Center > MUNICH | BUDAPEST | WARSAW > Phone: +44 (0) 863 801 69 > Web: www.virtual-call-center.eu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/91f5af44/attachment.html From nneul at mst.edu Mon Nov 3 16:45:34 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Mon, 03 Nov 2014 07:45:34 -0600 Subject: [Freeswitch-users] SCCP,buttons in phone configs In-Reply-To: <1414998769.410828978@f125.i.mail.ru> References: <1414998769.410828978@f125.i.mail.ru> Message-ID: <545786FE.2000207@mst.edu> Redial in particular requires dialplan support. In my setup, I have the redial extension set to "skinny-redial" and these two at the top of my dialplan: -- Nathan On 11/03/2014 01:12 AM, Nick Zaitsev wrote: > > Good day to you,maybe it is silly question,but, I actually can't understand how tel buttons can be set for sccp phones > in freeswitch. > For example, how can i set the redial button for Cisco IP Phone 7961 > > -- > Nick Zaitsev > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From joelewhite at gmail.com Mon Nov 3 17:54:37 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 3 Nov 2014 09:54:37 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: I will look into that Chris, thank you. I did set the inbound codec to force g711a + u and also did the same on outbound calls heading to our ITSP. On Tue, Oct 28, 2014 at 7:23 PM, Chris Tunbridge wrote: > Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup your > codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White wrote: > >> Also, just to mention. The CPU is never above 2% and the current load is >> about 20 simultaneous calls >> >> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >> wrote: >> >>> Is this an issue with transcoding? >>> >>> I am wondering how I lock it down so that transcoding only happens at a >>> minimum >>> >>> Also, would it be bad to completely remove G722 from the codec list? >>> >>> I think the issue is that our provider is sending G711 and all of the >>> Polycom Phones are trying to use G722 >>> >>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>> wrote: >>> >>>> I am having an issue where some calls start out fine and then go into >>>> slow motion (distorted slow speech) >>>> >>>> I noticed something in the logs stating >>>> >>>> Activating write resampler >>>> >>>> >>>> What does this mean and how do I avoid this in the future? >>>> >>>> >>>> >>>> Thank you in advance >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/8be849e7/attachment-0001.html From joelewhite at gmail.com Mon Nov 3 17:56:03 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 3 Nov 2014 09:56:03 -0500 Subject: [Freeswitch-users] Outbound Caller ID in a multi-Department environment In-Reply-To: References: Message-ID: Ok Chris, I thought that effective callerid was for internal (e.x. Intra Company Extension Number). I will try that today Thank you On Tue, Oct 28, 2014 at 7:25 PM, Chris Tunbridge wrote: > You need to set Effective Caller ID effective_caller_id_number and > effective_caller_id_name > > On Mon, Oct 20, 2014 at 10:26 AM, Joel White wrote: > >> 2014-10-20 12:22:08.594859 [NOTICE] switch_cpp.cpp:1328 Debug from >> gen_dir_user_xml.lua, generated XML: >> >> >>
>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >>
>> >> >> >> Michael, this is what I am dynamically generating using Postgres and >> Lua. I put the lines of code for the variables wanted in my outbound >> dialplan >> >> > data="effective_caller_id_number=${outbound_call >> er_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller >> _name}"/> >> >> >> I am still seeing the extension as the CID. What is missing here? >> >> This is why I asked if it was possible to write a lua script to do the db >> lookup and insert while progressing through the dialplan. >> >> >> >> What is missing? >> >> Also, is there a way to look at all variables set on a specific extension? >> >> >> >> >> >> On Fri, Oct 17, 2014 at 11:59 AM, Michael Collins >> wrote: >> >>> The bridge app doesn't care whether it's static XML or dynamically >>> generated. It only cares about the channel variables being populated. It >>> will be the same whether you make 100 calls all with the same OB CID or 100 >>> calls each with 100 different OB CID values. Just make sure that you're >>> setting the effective_caller_id_number (and _name, if necessary) variable. >>> >>> -MC >>> >>> >>> On Fri, Oct 17, 2014 at 8:33 AM, Joel White >>> wrote: >>> >>>> Will this work well when we are talking about perhaps 100 departments >>>> and 4000 users? >>>> >>>> I am trying to make it all run out of the database as changes can be >>>> made on the fly without disrupting call flow >>>> >>>> On Wed, Oct 15, 2014 at 2:53 PM, Steven Schoch < >>>> schoch+freeswitch.org at xwin32.com> wrote: >>>> >>>>> It should just work with the example dialplan, if you set the >>>>> variables. >>>>> In my default dialplan, in an outbound file, I have this extension: >>>>> >>>>> >>>>> >>>> expression="^\+?1?([2-9]\d\d[2-9]\d\d\ >>>>> d{4})$"> >>>>> >>>> data="effective_caller_id_number=${outbound_call >>>>> er_id_number}"/> >>>>> >>>> data="effective_caller_id_name=${outbound_caller >>>>> _name}"/> >>>>> >>>>> >>>> data="effective_callee_id_name=${cidlookup(1$1)} >>>>> "/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/gateway/$${default_provider}/1$1" >>>>> /> >>>>> >>>>> >>>>> >>>>> This sets the CID number to the outbound_caller_id_number, which is a >>>>> variable for each entry in the directory. A typical directory entry will >>>>> look like this: >>>>> >>>>> >>>>> >>>>> >>>>> ...user stuff here... >>>>> >>>>> >>>>> >>>> value="domestic,international,local"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="$${outbound_caller_name}"/> >>>>> >>>> value="$${outbound_caller_id}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Some phones have a different outbound_caller_id, but most use our main >>>>> line, which is set in vars.xml. >>>>> >>>>> -- >>>>> Steve >>>>> >>>>> >>>>> On Wed, Oct 15, 2014 at 9:06 AM, Joel White >>>>> wrote: >>>>> >>>>>> I am having an issue setting up CID for individual extensions >>>>>> >>>>>> >>>>>> I am running out of a PostgreSQL DB and have created a Users Table >>>>>> >>>>>> Using Lua I am dynamically generating the Directory >>>>>> >>>>>> I have a column in the table for CID, but have been unsuccessful in >>>>>> implementing CID for outbound calling >>>>>> >>>>>> >>>>>> I want the extension to display internally, but to add CID when >>>>>> calling outside of the system >>>>>> >>>>>> >>>>>> How can I implement this? And can it be done using Lua to >>>>>> dynamically lookup in the DB for each users CID? >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/5fc2b80d/attachment-0001.html From brian at freeswitch.org Mon Nov 3 18:42:56 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 3 Nov 2014 09:42:56 -0600 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: What rev are you on? How are you replicating this? Can you get a siptrace and logs and file a JIRA so we can get this bug fixed. Sounds like the codec is probably changing during the call, are you doing a hold/unhold operation? On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge wrote: > Joel is there any way you can test with one polycom set to g711u. > > This sounds to me like a 16k vs 8k transcoding issue, maybe setup your > codecs on the freeswitch side to only allow 8k g722 (if that's even > possible) > > On Tue, Oct 21, 2014 at 8:22 AM, Joel White wrote: > >> Also, just to mention. The CPU is never above 2% and the current load is >> about 20 simultaneous calls >> >> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >> wrote: >> >>> Is this an issue with transcoding? >>> >>> I am wondering how I lock it down so that transcoding only happens at a >>> minimum >>> >>> Also, would it be bad to completely remove G722 from the codec list? >>> >>> I think the issue is that our provider is sending G711 and all of the >>> Polycom Phones are trying to use G722 >>> >>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>> wrote: >>> >>>> I am having an issue where some calls start out fine and then go into >>>> slow motion (distorted slow speech) >>>> >>>> I noticed something in the logs stating >>>> >>>> Activating write resampler >>>> >>>> >>>> What does this mean and how do I avoid this in the future? >>>> >>>> >>>> >>>> Thank you in advance >>>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/23845996/attachment.html From joelewhite at gmail.com Mon Nov 3 19:44:50 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 3 Nov 2014 11:44:50 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: I will work on getting a sip trace. I am seeing this happen when monitoring the system. The calls sound fine and then it sounds like the audio slows to a crawl. It does not appear to happen regularly and I have seen it happen on individual calls and on conferences. The conferences have external DIDs for outside participants. The internal extensions dial a 5 digit extension. On Mon, Nov 3, 2014 at 10:42 AM, Brian West wrote: > What rev are you on? How are you replicating this? Can you get a siptrace > and logs and file a JIRA so we can get this bug fixed. Sounds like the > codec is probably changing during the call, are you doing a hold/unhold > operation? > > On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge > wrote: > >> Joel is there any way you can test with one polycom set to g711u. >> >> This sounds to me like a 16k vs 8k transcoding issue, maybe setup your >> codecs on the freeswitch side to only allow 8k g722 (if that's even >> possible) >> >> On Tue, Oct 21, 2014 at 8:22 AM, Joel White wrote: >> >>> Also, just to mention. The CPU is never above 2% and the current load >>> is about 20 simultaneous calls >>> >>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>> wrote: >>> >>>> Is this an issue with transcoding? >>>> >>>> I am wondering how I lock it down so that transcoding only happens at a >>>> minimum >>>> >>>> Also, would it be bad to completely remove G722 from the codec list? >>>> >>>> I think the issue is that our provider is sending G711 and all of the >>>> Polycom Phones are trying to use G722 >>>> >>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>> wrote: >>>> >>>>> I am having an issue where some calls start out fine and then go into >>>>> slow motion (distorted slow speech) >>>>> >>>>> I noticed something in the logs stating >>>>> >>>>> Activating write resampler >>>>> >>>>> >>>>> What does this mean and how do I avoid this in the future? >>>>> >>>>> >>>>> >>>>> Thank you in advance >>>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/1bcf1447/attachment.html From joelewhite at gmail.com Mon Nov 3 19:49:16 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 3 Nov 2014 11:49:16 -0500 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit) Current version on the server On Mon, Nov 3, 2014 at 11:44 AM, Joel White wrote: > I will work on getting a sip trace. I am seeing this happen when > monitoring the system. The calls sound fine and then it sounds like the > audio slows to a crawl. It does not appear to happen regularly and I have > seen it happen on individual calls and on conferences. > > The conferences have external DIDs for outside participants. The internal > extensions dial a 5 digit extension. > > > > > On Mon, Nov 3, 2014 at 10:42 AM, Brian West wrote: > >> What rev are you on? How are you replicating this? Can you get a siptrace >> and logs and file a JIRA so we can get this bug fixed. Sounds like the >> codec is probably changing during the call, are you doing a hold/unhold >> operation? >> >> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >> wrote: >> >>> Joel is there any way you can test with one polycom set to g711u. >>> >>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup your >>> codecs on the freeswitch side to only allow 8k g722 (if that's even >>> possible) >>> >>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>> wrote: >>> >>>> Also, just to mention. The CPU is never above 2% and the current load >>>> is about 20 simultaneous calls >>>> >>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>> wrote: >>>> >>>>> Is this an issue with transcoding? >>>>> >>>>> I am wondering how I lock it down so that transcoding only happens at >>>>> a minimum >>>>> >>>>> Also, would it be bad to completely remove G722 from the codec list? >>>>> >>>>> I think the issue is that our provider is sending G711 and all of the >>>>> Polycom Phones are trying to use G722 >>>>> >>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>> wrote: >>>>> >>>>>> I am having an issue where some calls start out fine and then go into >>>>>> slow motion (distorted slow speech) >>>>>> >>>>>> I noticed something in the logs stating >>>>>> >>>>>> Activating write resampler >>>>>> >>>>>> >>>>>> What does this mean and how do I avoid this in the future? >>>>>> >>>>>> >>>>>> >>>>>> Thank you in advance >>>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/db0e4b1d/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 3 20:00:04 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 3 Nov 2014 11:00:04 -0600 Subject: [Freeswitch-users] Issues with rtp audio going in slow motion In-Reply-To: References: Message-ID: This should be handled in JIRA not the mailing list. If problem or issue seems prudent in the subject, think JIRA http://jira.freeswitch.org That way you can post logs etc and not flood the list with help questions. On Mon, Nov 3, 2014 at 10:49 AM, Joel White wrote: > FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git > ae069dc 2014-09-29 19:49:48Z 64bit) > > Current version on the server > > On Mon, Nov 3, 2014 at 11:44 AM, Joel White wrote: > >> I will work on getting a sip trace. I am seeing this happen when >> monitoring the system. The calls sound fine and then it sounds like the >> audio slows to a crawl. It does not appear to happen regularly and I have >> seen it happen on individual calls and on conferences. >> >> The conferences have external DIDs for outside participants. The >> internal extensions dial a 5 digit extension. >> >> >> >> >> On Mon, Nov 3, 2014 at 10:42 AM, Brian West wrote: >> >>> What rev are you on? How are you replicating this? Can you get a >>> siptrace and logs and file a JIRA so we can get this bug fixed. Sounds >>> like the codec is probably changing during the call, are you doing a >>> hold/unhold operation? >>> >>> On Tue, Oct 28, 2014 at 6:23 PM, Chris Tunbridge >>> wrote: >>> >>>> Joel is there any way you can test with one polycom set to g711u. >>>> >>>> This sounds to me like a 16k vs 8k transcoding issue, maybe setup your >>>> codecs on the freeswitch side to only allow 8k g722 (if that's even >>>> possible) >>>> >>>> On Tue, Oct 21, 2014 at 8:22 AM, Joel White >>>> wrote: >>>> >>>>> Also, just to mention. The CPU is never above 2% and the current load >>>>> is about 20 simultaneous calls >>>>> >>>>> On Tue, Oct 21, 2014 at 10:16 AM, Joel White >>>>> wrote: >>>>> >>>>>> Is this an issue with transcoding? >>>>>> >>>>>> I am wondering how I lock it down so that transcoding only happens at >>>>>> a minimum >>>>>> >>>>>> Also, would it be bad to completely remove G722 from the codec list? >>>>>> >>>>>> I think the issue is that our provider is sending G711 and all of the >>>>>> Polycom Phones are trying to use G722 >>>>>> >>>>>> On Mon, Oct 20, 2014 at 11:43 AM, Joel White >>>>>> wrote: >>>>>> >>>>>>> I am having an issue where some calls start out fine and then go >>>>>>> into slow motion (distorted slow speech) >>>>>>> >>>>>>> I noticed something in the logs stating >>>>>>> >>>>>>> Activating write resampler >>>>>>> >>>>>>> >>>>>>> What does this mean and how do I avoid this in the future? >>>>>>> >>>>>>> >>>>>>> >>>>>>> Thank you in advance >>>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/404ab3fa/attachment.html From mvar78 at gmail.com Mon Nov 3 20:20:00 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 3 Nov 2014 18:20:00 +0100 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> Message-ID: <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> Hi Ken! Thank you for your answer, but I have another question regarding this....can this variable be set into a LUA script or is a XML global variable? I'm trying to set this option this way but the value that nibble is considering is the XML (zero) and drop the call session:execute("set","nobal_amt=-10") Thank you so much! Max Il giorno 01/nov/2014, alle ore 15:10, Ken Rice ha scritto: > you just allow a negative balance for a post paid customer > > Ken > Sent from my iPad > >> On Nov 1, 2014, at 8:30, Massimo Varriale wrote: >> >> Hi All, >> is possible to create a scenario in which I have a postpaid customer in which the credit can go below zero but no more than a maximum value allowed? >> The wiki talks about the feature of having a postpaid customer, but there are no example regarding this feature. But on prepaid there are many and many example, thank for this, great! >> >> BTW, currently I can create a prepaid customer and as soon as credit reach 0 the call is dropped correctly, but how can I implement a postpaid customer? >> >> Thank you so much >> Max >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From avi at avimarcus.net Mon Nov 3 20:35:49 2014 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 3 Nov 2014 17:35:49 +0000 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> Message-ID: <0000014976ba20a3-a7ec3ec2-9c92-4144-b100-6dbfa40951e9-000000@email.amazonses.com> The old Wiki says it's a nibblebill configuration variable: In your conf/autoload_configs/nibblebill.conf.xml add something like this: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/709164c2/attachment.html From mvar78 at gmail.com Mon Nov 3 20:45:47 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 3 Nov 2014 18:45:47 +0100 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <0000014976ba20a3-a7ec3ec2-9c92-4144-b100-6dbfa40951e9-000000@email.amazonses.com> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> <0000014976ba20a3-a7ec3ec2-9c92-4144-b100-6dbfa40951e9-000000@email.amazonses.com> Message-ID: Yes, that's true. My XML parameter is using just this parameters, however I'm trying to find a way to bill a call based on a rate and reduce the credit accordly. In my Lua script I'm trying to use this syntax to build the bridge string: So, no way to set this variable dinamically? Thank you Max Il giorno 03/nov/2014, alle ore 18:35, Avi Marcus ha scritto: > The old Wiki says it's a nibblebill configuration variable: > > In your conf/autoload_configs/nibblebill.conf.xml add something like this: > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/41a71888/attachment-0001.html From mvar78 at gmail.com Mon Nov 3 20:48:33 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 3 Nov 2014 18:48:33 +0100 Subject: [Freeswitch-users] Fwd: [Lua Script] Conditional Remapping of Hangup Cause References: Message-ID: Hi Guys! Does anyone can point me the right direction to work on changing remote Hangup Cause and give my FS customer another Hangup Cause? I'm lost :( Thank you so much Max ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Hi, I'm creating a Lua script for managing different call scenario. At the moment I'm stuck with a problem regarding the interoperability between Freeswitch and some potential customer. The problem regards the hangup cause that Freeswitch is sending to their softswitch, to allow a their automatic rerouting based on some Hangup Cause. I got a request to send different Hangup Cause, so the Customer switch can automatically reroute call to another carrier. Here is the explaination The scenario is: All the endpoints are using a Public IP and no NAT is involved in these steps. CUSTOMER ------------------> MY_FREESWITCH -----------------> REMOTE_CARRIER What happen now: The customer can send calls. On My freeswitch I will route calls based on Dialled Prefix to a Remote Carrier. In my Lua script there is a conditional IF to check for remote disconnect cause, however it seems that the Release Cause is already sent (CALL_REJECTED) when I'm checking with my IF and so, even if the cause has been already sent it's sent again but what "win" is the first hangup sent. What should be: If the remote carrier will send me a CALL_REJECTED (isdn 21) I need to send to the remote customer another Hangup Cause for example: NORMAL_CIRCUIT_CONGESTION (isdn 34). How to implement that? I can create this remapping using a Diaplan XML configuration, but as I'm using a Lua script to manage the call, I found that is a Lua task to deal with the Hangup Cause, the dialplan rules are ignored. The "telephony" part of the Lua script is working, what I'm trying to change is the release cause to make my customer happy as with a different release cause he can route traffic correctly and I will not loose my reputation :) Below there is my Lua script (just the routing part) Thank you so much! Cheers Max > function string.starts(String,Start) > return string.sub(String,1,string.len(Start))==Start > end > > session:execute("set","hangup_after_bridge=true") > called_number = session:getVariable("destination_number") > > > while session:answered()==false and session:ready()==true do > > if string.starts(called_number, "393") then > -- ITALY MOBILE > route_dial_string = "sofia/gateway/REMOTE_CARRIER/" ..called_number > else > -- OTHER DESTINATIONS > route_dial_string = "sofia/gateway/OTHER_CARRIER/" ..called_number > end > > session:execute("bridge",""..route_dial_string.."") > > > -- ATTEMPT TO REMAP RELEASE CAUSE > if ( session:getVariable("last_bridge_hangup_cause") == "CALL_REJECTED" ) then > session:hangup("NORMAL_CIRCUIT_CONGESTION"); > end > > > if session:ready()==false then > session:hangup(); > > > end > > > session:hangup(); > freeswitch.consoleLog("info", "That's all folks!"); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/164c1fa4/attachment.html From mariogasparoni at gmail.com Mon Nov 3 20:59:32 2014 From: mariogasparoni at gmail.com (Mario Gasparoni Jr) Date: Mon, 3 Nov 2014 15:59:32 -0200 Subject: [Freeswitch-users] Software/Hardware interoperability Message-ID: Hello guys, I am trying to use FreeSWITCH as a conference server for group calls. I'm using mod_conference, and I know it works fine with multiple SIP clients (for testing purposes I use Ekiga and Jitsi on both Windows and Linux). Audio is mixed individually for each user, and video is switched based on the active talker, and the same video is sent to all clients. The problem shows up when I join the group call using a Polycom hardware. If I configure FreeSWITCH to support H.263 only, it works fine, my Polycom can display the video sent by FreeSWITCH, including when the active talker is itself. The other SIP clients work fine as well. When I configure FreeSWITCH to support H.264 only, the video switching only works when Polycom is the active talker, otherwise all the users just see the last video frame sent by Polycom. If I do a peer-to-peer call between the Polycom and a SIP client, both configured to use H.264, everything works just fine, which points me that the problem could be related to the transition between viewing a video stream with a given resolution and viewing another video stream with a different resolution, or something related to the non-continuity of video packets timestamp. Is there anybody out there that have experienced similar issues? Do you guys have any suggestion or anything that could help me to make it work properly? Thanks in advance. -- Att. M?rio Gasparoni. www.inf.ufrgs.br/~mcgjunior -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/e5347819/attachment.html From fvillarroel at yahoo.com Mon Nov 3 21:23:49 2014 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Mon, 3 Nov 2014 10:23:49 -0800 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> <0000014976ba20a3-a7ec3ec2-9c92-4144-b100-6dbfa40951e9-000000@email.amazonses.com> Message-ID: <1415039029.95746.YahooMailNeo@web162004.mail.bf1.yahoo.com> I define nobal_amt variable for each user like this: ...... So in my dialplan i have: Regards On Monday, November 3, 2014 2:48 PM, Massimo Varriale wrote: Yes, that's true. My XML parameter is using just this parameters, however I'm trying to find a way to bill a call based on a rate and reduce the credit accordly. In my Lua script I'm trying to use this syntax to build the bridge string: So, no way to set this variable dinamically? Thank you Max Il giorno 03/nov/2014, alle ore 18:35, Avi Marcus ha scritto: The old Wiki says it's a nibblebill configuration variable: > > >In your conf/autoload_configs/nibblebill.conf.xml add something like this: > _________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://confluence.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/bd210dbb/attachment-0001.html From joelewhite at gmail.com Tue Nov 4 00:36:26 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 3 Nov 2014 16:36:26 -0500 Subject: [Freeswitch-users] PostgreSQL - p_key In-Reply-To: References: <543F3C95.5050807@quentustech.com> Message-ID: Is there another datatype that would be more suitable. I plan to have massive amounts of traffic. Would the time datatype be more suitable? On Mon, Oct 20, 2014 at 12:26 PM, Joel White wrote: > Thank you Chris > > On Fri, Oct 17, 2014 at 2:08 PM, Chris Tunbridge > wrote: > >> Serial is identical to mysql's Auto Increment, so i'm going to say yes. >> >> On Fri, Oct 17, 2014 at 9:36 AM, Joel White wrote: >> >>> Chris and William, >>> >>> Can those added PKs have a data type of SERIAL? >>> >>> >>> >>> On Wed, Oct 15, 2014 at 11:33 PM, William King < >>> william.king at quentustech.com> wrote: >>> >>>> Yes, you can add columns to those tables with no issues. Try it and >>>> see. >>>> >>>> William King >>>> Senior Engineer >>>> Quentus Technologies, INC >>>> 1037 NE 65th St Suite 273 >>>> Seattle, WA 98115 >>>> Main: (877) 211-9337 >>>> Office: (206) 388-4772 >>>> Cell: (253) 686-5518william.king at quentustech.com >>>> >>>> On 10/8/14 9:44 AM, Joel White wrote: >>>> >>>> Is it feasible to add Primary Keys to all FreeSWITCH tables or would >>>> this break some functionality? >>>> >>>> I am asking because I am working on a multi-master replication setup >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/a7eff8bc/attachment.html From mike at jerris.com Tue Nov 4 00:50:50 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 3 Nov 2014 16:50:50 -0500 Subject: [Freeswitch-users] PostgreSQL - p_key In-Reply-To: References: <543F3C95.5050807@quentustech.com> Message-ID: I've used bigint with a default of a sequence in the past to good results. make sure to add the option to not recreate the tables so we don't mess up your fields when we change scheme and when upgrading review for schema changes and make them manually On Monday, November 3, 2014, Joel White wrote: > Is there another datatype that would be more suitable. I plan to have > massive amounts of traffic. Would the time datatype be more suitable? > > > > On Mon, Oct 20, 2014 at 12:26 PM, Joel White > wrote: > >> Thank you Chris >> >> On Fri, Oct 17, 2014 at 2:08 PM, Chris Tunbridge > > wrote: >> >>> Serial is identical to mysql's Auto Increment, so i'm going to say yes. >>> >>> On Fri, Oct 17, 2014 at 9:36 AM, Joel White >> > wrote: >>> >>>> Chris and William, >>>> >>>> Can those added PKs have a data type of SERIAL? >>>> >>>> >>>> >>>> On Wed, Oct 15, 2014 at 11:33 PM, William King < >>>> william.king at quentustech.com >>>> > wrote: >>>> >>>>> Yes, you can add columns to those tables with no issues. Try it and >>>>> see. >>>>> >>>>> William King >>>>> Senior Engineer >>>>> Quentus Technologies, INC >>>>> 1037 NE 65th St Suite 273 >>>>> Seattle, WA 98115 >>>>> Main: (877) 211-9337 >>>>> Office: (206) 388-4772 >>>>> Cell: (253) 686-5518william.king at quentustech.com >>>>> >>>>> On 10/8/14 9:44 AM, Joel White wrote: >>>>> >>>>> Is it feasible to add Primary Keys to all FreeSWITCH tables or would >>>>> this break some functionality? >>>>> >>>>> I am asking because I am working on a multi-master replication setup >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/1e303ff8/attachment-0001.html From krice at freeswitch.org Tue Nov 4 02:42:07 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 03 Nov 2014 23:42:07 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! Message-ID: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> New Post on freeswitch.org from krice387 check it out at http://ift.tt/1rUJmfP FreeSWITCH 1.4.13 Released!! The FreeSWITCH Team is happy to announce the release of FreeSWITCH 1.4.13!!!! Tarball is available at http://ift.tt/1rUJls8 as well as packages for Debian and Centos6 in their respective repositories! This Release includes: Fix SegFault in switch_core_media.c ? http://ift.tt/1ocyP4k Updates for Xcode 6 on OSX Mavericks and Yosemite ? http://ift.tt/1rUJlsa Allow sub millisecond resolution for option ping times ? http://ift.tt/1rUJlsc Add ability to log commands executed in mod_xml_rpc Fix error when resuming a call on hold ? http://ift.tt/1rUJmMJ Fix missing Notify message record-route tag ? http://ift.tt/1ocyRZS Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load Improve SIP OPTIONS ping generation by distributing them across an interval ? http://ift.tt/1rUJlsi variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X?s in the CDR (useful for filtering things like CreditCard numbers) ? http://ift.tt/1ocyS00 fix leak of nua handle due to reference counting. Effects all calls with auth/challenge on INVITE as well as several other minor tweaks and changes There are also several other minor changes and bug fixes in this release! Along with this release we had to fix a small error in the main FreeSWITCH GIT repository on the v1.4 branch. On your next pull of the branch you may see a merge error. To resolve this error from your local copy of the git repo while on the v1.4 branch issue the following command:git reset --hard origin/v1.4 Please note, if you have local patches you will want to rebase from the main FreeSWITCH git repository to resolve the issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141103/bb51f0e0/attachment.html From krice at freeswitch.org Tue Nov 4 06:26:45 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 04 Nov 2014 03:26:45 +0000 Subject: [Freeswitch-users] Good News Everyone! We can finally use HTML5, Oh yeah we already were! Message-ID: <54584775a44b1_5bdb51d330185c1@ip-10-156-213-204.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/1xVsqdv Good News Everyone! We can finally use HTML5, Oh yeah we already were! After almost 15 years, we can finally get started on HTML6! Maybe it?ll be done in time for FreeSWITCH 3.0 http://ift.tt/1zGJkBV ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/8064799f/attachment.html From varghesepaul87 at gmail.com Tue Nov 4 10:02:22 2014 From: varghesepaul87 at gmail.com (Varghese Paul) Date: Tue, 4 Nov 2014 12:32:22 +0530 Subject: [Freeswitch-users] To decypt webrtc audio In-Reply-To: References: Message-ID: Hi Akio, Yes the use case is to check the audio quality. Regards Varghese Paul On Sun, Nov 2, 2014 at 2:58 AM, Ben Langfeld wrote: > Ah, this is not what I understood from the original question. My apologies > for the detour. > > On 31 October 2014 12:38, Varghese Paul wrote: > >> Hi Ben, >> >> I am looking for a tool like wireshark to decrypt the SRTP packets and >> listen to the audio. RTPengine as you said we can bridge two channels and >> we can't use it as a tool for decrypting SRTP traffic, >> >> Regards >> >> On Fri, Oct 31, 2014 at 7:25 PM, Carlos Ruiz D?az < >> carlos.ruizdiaz at gmail.com> wrote: >> >>> I understand that this is not what he is intending. Why would he need >>> yet another rtp bridge if FS already does exactly that. >>> On Oct 31, 2014 7:36 AM, "Ben Langfeld" wrote: >>> >>>> The way I use rtpengine, it rewrites SDP between the browser and >>>> Asterisk 1.4, terminating DTLS and ICE. Between the browser and rtpengine >>>> is RTP/SAVPF, while between rtpengine and Asterisk is RTP/AVP. rtpengine >>>> generates its keys in what I assume to be the same way the browsers do. >>>> >>>> On 31 October 2014 10:51, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> The tool from pjsip mentioned above requires the keys for the srtp >>>>> decryption. Does the one you mention do it differently? >>>>> >>>>> >>>>> On Friday, October 31, 2014, Ben Langfeld wrote: >>>>> >>>>>> Anthony, I think I'm not understanding your concern. Could you >>>>>> explain what you mean by logging the keys? >>>>>> >>>>>> On 30 October 2014 21:01, Anthony Minessale < >>>>>> anthony.minessale at gmail.com> wrote: >>>>>> >>>>>>> The bigger question is how to log the keys without compromising the >>>>>>> point of why its all encrypted. >>>>>>> It would take some careful consideration. >>>>>>> >>>>>>> >>>>>>> On Thu, Oct 30, 2014 at 12:32 PM, Ben Langfeld >>>>>>> wrote: >>>>>>> >>>>>>>> rtpengine does indeed allow decryption of DTLS encrypted RTP. I use >>>>>>>> it in front of Asterisk 1.4 for exactly that (and ICE resolution). >>>>>>>> >>>>>>>> On 30 October 2014 14:55, Carlos Ruiz D?az < >>>>>>>> carlos.ruizdiaz at gmail.com> wrote: >>>>>>>> >>>>>>>>> rtpengine doesn't provide any mean to do any kind of decryption. >>>>>>>>> The keys, however, are easily accessible if you are willing to modify the >>>>>>>>> original software to create such functionality. >>>>>>>>> >>>>>>>>> Regards, >>>>>>>>> Carlos >>>>>>>>> >>>>>>>>> On Thu, Oct 30, 2014 at 10:47 AM, Anthony Minessale < >>>>>>>>> anthony.minessale at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> This is a philosophical debate as to weather or not making the >>>>>>>>>> key available somewhere it can be obtained from nullifies the intended >>>>>>>>>> security. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Thu, Oct 30, 2014 at 5:33 AM, Ben Langfeld >>>>>>>>> > wrote: >>>>>>>>>> >>>>>>>>>>> Look into rtpengine and perhaps Kamailio to control it. >>>>>>>>>>> >>>>>>>>>>> On 30 October 2014 04:41, Varghese Paul < >>>>>>>>>>> varghesepaul87 at gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi all, >>>>>>>>>>>> >>>>>>>>>>>> We are using freeswitch for handling webrtc calls and the >>>>>>>>>>>> negotiation in DTLS mode. >>>>>>>>>>>> >>>>>>>>>>>> We are looking for a tool to decrypt the traffic between the >>>>>>>>>>>> webrtc client and freeswitch server. I know there is already a tool >>>>>>>>>>>> pcaputils from PJSIP for decrypting the SRTP traffic. Since we are using in >>>>>>>>>>>> DTLS mode we can't use this. >>>>>>>>>>>> >>>>>>>>>>>> Can any one suggest any tool/method to decrypt the webrtc >>>>>>>>>>>> audio? >>>>>>>>>>>> >>>>>>>>>>>> Regards >>>>>>>>>>>> >>>>>>>>>>>> Varghese Paul >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting at freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>>>>>>> >>>>>>>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>>>>>>> * >>>>>>>>>> >>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting at freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://confluence.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Carlos >>>>>>>>> http://caruizdiaz.com >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://confluence.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>>>> >>>>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>>>> http://twitter.com/FreeSWITCH >>>>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>>>> * >>>>>>> >>>>>>> ClueCon Weekly Development Call >>>>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? >>>>> >>>>> ? http://freeswitch.org/ ? http://cluecon.com/ ? >>>>> http://twitter.com/FreeSWITCH >>>>> ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ? sip:888 at conference.freeswitch.org ? +19193869900 >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/7cc08782/attachment-0001.html From moises.silva at gmail.com Tue Nov 4 11:05:14 2014 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 4 Nov 2014 03:05:14 -0500 Subject: [Freeswitch-users] freeswitch + openr2 In-Reply-To: <20141029013738.10136t0hse0z8182@webmail.tiendaip.mx> References: <20141029013738.10136t0hse0z8182@webmail.tiendaip.mx> Message-ID: Hi Leon, hope things are going good for you in Mexico :) On Wed, Oct 29, 2014 at 2:37 AM, wrote: > Dear list: > > I am facing a problem when trying to compile freeswitch, freetdm + openr2. > > I followed the wiki on how to do it, I was able to use the old 1.2 > openr2 version. > > A. If I use the version 1.2 from the svn trunk and the cmake command > You should use github master: https://github.com/moises-silva/openr2 I can't recall if there's differences though, but svn is definitely not the best option now days, I might have done changes in git that are not in svn anymore (I'm surprised svn still works, I thought I had removed it) Try using master from github, "cd build/ && cmake .. && make install" or something like that Then when freetdm is configured, if it still does not detect openr2, please share config.log using pastebin (it's located in the freetdm/ directory, is a file produced when running ./configure) Moy Moises Silva Manager, Software Engineering Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, Ontario, Canada L3R 9R6 T +1 905 474 1990 x128 | toll-free in North America +1 800 388-2475 | F +1 905 474 9223 www.sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/2d16a4a4/attachment.html From moises.silva at gmail.com Tue Nov 4 11:16:17 2014 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 4 Nov 2014 03:16:17 -0500 Subject: [Freeswitch-users] Embedded Freeswitch In-Reply-To: References: <005c01cfd5bf$ce3c93a0$6ab5bae0$@comcast.net> Message-ID: Hi Guillermo, On Mon, Sep 22, 2014 at 12:55 PM, Guillermo Ruiz Camauer < grcamauer at gmail.com> wrote: > Andrew, > > Thanks for the response. I am wondering if there are any examples using > C/C++. > You can have a look at the fscomm/ directory in FreeSWITCH git repo, then peak at fshost.cpp ( https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/fscomm/fshost.cpp ) Pay particular attention to the FSHost::run() method calling the following functions: switch_core_init() switch_core_init_and_modload() switch_core_runtime_loop() That's the gist of it to get started. In my opinion most of the time is better to launch a lean freeswitch process (with only the modules you need in your app) in the background and control it using ESL Cheers Moy Moises Silva Manager, Software Engineering Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, Ontario, Canada L3R 9R6 T +1 905 474 1990 x128 | toll-free in North America +1 800 388-2475 | F +1 905 474 9223 www.sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/df58c17a/attachment.html From mkvonarx at gmail.com Tue Nov 4 11:24:45 2014 From: mkvonarx at gmail.com (Markus von Arx) Date: Tue, 4 Nov 2014 09:24:45 +0100 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> Message-ID: Hi Ken Thanks for the detailed release announcement! Any information about why the downloads are so huge for the 1.4.13 release? freeswitch-1.4.12.tar.bz2 was 37MB, freeswitch-1.4.13.tar.bz2 now is 442MB. The tar file contains 263MB of "rpmbuild", 168MB of "src_dist" and 126MB of "libs". Markus 2014-11-04 0:42 GMT+01:00 Ken Rice : > New Post on freeswitch.org from krice387 > check it out at http://ift.tt/1rUJmfP > FreeSWITCH 1.4.13 Released!! > > The FreeSWITCH Team is happy to announce the release of FreeSWITCH > 1.4.13!!!! > > Tarball is available at http://ift.tt/1rUJls8 as well as packages for > Debian and Centos6 in their respective repositories! > > This Release includes: > > - Fix SegFault in switch_core_media.c ? http://ift.tt/1ocyP4k > - Updates for Xcode 6 on OSX Mavericks and Yosemite ? > http://ift.tt/1rUJlsa > - Allow sub millisecond resolution for option ping times ? > http://ift.tt/1rUJlsc > - Add ability to log commands executed in mod_xml_rpc > - Fix error when resuming a call on hold ? http://ift.tt/1rUJmMJ > - Fix missing Notify message record-route tag ? http://ift.tt/1ocyRZS > - Add new hard_mute control to allow apps to request low level mute > e.g. from the rtp stack level. Its used in mod_conference to avoid reading > audio while muted and possibly reduce some transcoding load > - Improve SIP OPTIONS ping generation by distributing them across an > interval ? http://ift.tt/1rUJlsi > - variable digits_dialed_filter to set regular expressions with () > captures and anything matched will be replaced with X?s in the CDR (useful > for filtering things like CreditCard numbers) ? http://ift.tt/1ocyS00 > - fix leak of nua handle due to reference counting. Effects all calls > with auth/challenge on INVITE > - as well as several other minor tweaks and changes > > There are also several other minor changes and bug fixes in this release! > > Along with this release we had to fix a small error in the main FreeSWITCH > GIT repository on the v1.4 branch. On your next pull of the branch you may > see a merge error. To resolve this error from your local copy of the git > repo while on the v1.4 branch issue the following command: > git reset --hard origin/v1.4 > Please note, if you have local patches you will want to rebase from the > main FreeSWITCH git repository to resolve the issue. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/98511902/attachment.html From balazs.sandor at virtual-call-center.hu Tue Nov 4 12:52:16 2014 From: balazs.sandor at virtual-call-center.hu (=?UTF-8?B?U8OhbmRvciBCYWzDoXpz?=) Date: Tue, 4 Nov 2014 10:52:16 +0100 Subject: [Freeswitch-users] mod_verto unwanted profile shutdown In-Reply-To: <1853D18C-7264-45FF-9B17-29487807EB48@jerris.com> References: <1853D18C-7264-45FF-9B17-29487807EB48@jerris.com> Message-ID: It not worked. Same issue 2014-11-04 10:18:09.731503 [CONSOLE] sofia_presence.c:1618 Event Thread Started freeswitch at default> verto status Name Type Data State ================================================================================================= mine profile ws:192.168.1.33:8081 RUNNING mine profile wss:192.168.1.33:8082 RUNNING ================================================================================================= 1 profile , 0 clients ... 2014-11-04 10:23:00.051524 [INFO] mod_verto.c:4524 profile mine shutdown, Waiting for 0 threads 2014-11-04 10:23:00.051524 [INFO] mod_verto.c:4535 mine Thread ending ?dv?zlettel: S?ndor Bal?zs Szoftver fejleszt? Virtual Call Center MUNICH | BUDAPEST | WARSAW Telefon: +36 1 999 7400 Web: www.virtual-call-center.hu 2014-11-03 14:40 GMT+01:00 Michael Jerris : > Does it do the same thing if you bind to a real ip instead of 0.0.0.0? > > On Nov 3, 2014, at 4:25 AM, Bal?zs S?ndor < > balazs.sandor at virtual-call-center.eu> wrote: > > Hi all. > > I'm testing mod_verto. > After a while (5-10 minutes) it always disconnecting. > I do nothing with FS (its sleeping) > > It seems to be normal shutdown > 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert > specified > 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to > 0.0.0.0:8081 > 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to > 0.0.0.0:8082 > -- > 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, > Waiting for 0 threads > 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending > > I attached a detailed log > > Best regards, > > Bal?zs S?NDOR > Software Developer > > > > Virtual Call Center > MUNICH | BUDAPEST > | WARSAW > > Phone: +44 (0) 863 801 69 > Web: www.virtual-call-center.eu > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/8754011b/attachment-0001.html From ssinyagin at gmail.com Tue Nov 4 15:28:47 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 4 Nov 2014 13:28:47 +0100 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: Dear Matsumoto-san, I think it will be easier if you write in Japanese, then it will be clear how we could help. I know a few Japanese-speaking colleagues who may help in communicating. On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > Hello > > I have two issues. > > I am writing in Perl. > > While 2 people are talking in a conference room, the one person want to > play the sound. > > In "Caller-Username", can you get useless. > I have tried the above but, Member: it will not become a *** not found.. > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > I want to play the above in the background > It can not play in the next program. > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav); > > $ con-> bgapi ($ api_cmd); > > Best regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/b1459925/attachment.html From steveayre at gmail.com Tue Nov 4 15:57:11 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 4 Nov 2014 12:57:11 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> Message-ID: Looks like among other things rpmbuild and src_dist contain full sets of tarballs of v1.4.12 On 4 November 2014 08:24, Markus von Arx wrote: > Hi Ken > > Thanks for the detailed release announcement! > > Any information about why the downloads are so huge for the 1.4.13 > release? freeswitch-1.4.12.tar.bz2 was 37MB, freeswitch-1.4.13.tar.bz2 now > is 442MB. The tar file contains 263MB of "rpmbuild", 168MB of "src_dist" > and 126MB of "libs". > > Markus > > > 2014-11-04 0:42 GMT+01:00 Ken Rice : > >> New Post on freeswitch.org from krice387 >> check it out at http://ift.tt/1rUJmfP >> FreeSWITCH 1.4.13 Released!! >> >> The FreeSWITCH Team is happy to announce the release of FreeSWITCH >> 1.4.13!!!! >> >> Tarball is available at http://ift.tt/1rUJls8 as well as packages for >> Debian and Centos6 in their respective repositories! >> >> This Release includes: >> >> - Fix SegFault in switch_core_media.c ? http://ift.tt/1ocyP4k >> - Updates for Xcode 6 on OSX Mavericks and Yosemite ? >> http://ift.tt/1rUJlsa >> - Allow sub millisecond resolution for option ping times ? >> http://ift.tt/1rUJlsc >> - Add ability to log commands executed in mod_xml_rpc >> - Fix error when resuming a call on hold ? http://ift.tt/1rUJmMJ >> - Fix missing Notify message record-route tag ? http://ift.tt/1ocyRZS >> - Add new hard_mute control to allow apps to request low level mute >> e.g. from the rtp stack level. Its used in mod_conference to avoid reading >> audio while muted and possibly reduce some transcoding load >> - Improve SIP OPTIONS ping generation by distributing them across an >> interval ? http://ift.tt/1rUJlsi >> - variable digits_dialed_filter to set regular expressions with () >> captures and anything matched will be replaced with X?s in the CDR (useful >> for filtering things like CreditCard numbers) ? http://ift.tt/1ocyS00 >> - fix leak of nua handle due to reference counting. Effects all calls >> with auth/challenge on INVITE >> - as well as several other minor tweaks and changes >> >> There are also several other minor changes and bug fixes in this release! >> >> Along with this release we had to fix a small error in the main >> FreeSWITCH GIT repository on the v1.4 branch. On your next pull of the >> branch you may see a merge error. To resolve this error from your local >> copy of the git repo while on the v1.4 branch issue the following command: >> git reset --hard origin/v1.4 >> Please note, if you have local patches you will want to rebase from the >> main FreeSWITCH git repository to resolve the issue. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/f363c91d/attachment.html From brian at freeswitch.org Tue Nov 4 16:23:09 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 4 Nov 2014 07:23:09 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> Message-ID: https://freeswitch.org/jira/browse/FS-6959 Ken will get this corrected today. On Tue, Nov 4, 2014 at 6:57 AM, Steven Ayre wrote: > Looks like among other things rpmbuild and src_dist contain full sets of > tarballs of v1.4.12 > > On 4 November 2014 08:24, Markus von Arx wrote: > >> Hi Ken >> >> Thanks for the detailed release announcement! >> >> Any information about why the downloads are so huge for the 1.4.13 >> release? freeswitch-1.4.12.tar.bz2 was 37MB, freeswitch-1.4.13.tar.bz2 now >> is 442MB. The tar file contains 263MB of "rpmbuild", 168MB of "src_dist" >> and 126MB of "libs". >> >> Markus >> >> >> 2014-11-04 0:42 GMT+01:00 Ken Rice : >> >>> New Post on freeswitch.org from krice387 >>> check it out at http://ift.tt/1rUJmfP >>> FreeSWITCH 1.4.13 Released!! >>> >>> The FreeSWITCH Team is happy to announce the release of FreeSWITCH >>> 1.4.13!!!! >>> >>> Tarball is available at http://ift.tt/1rUJls8 as well as packages for >>> Debian and Centos6 in their respective repositories! >>> >>> This Release includes: >>> >>> - Fix SegFault in switch_core_media.c ? http://ift.tt/1ocyP4k >>> - Updates for Xcode 6 on OSX Mavericks and Yosemite ? >>> http://ift.tt/1rUJlsa >>> - Allow sub millisecond resolution for option ping times ? >>> http://ift.tt/1rUJlsc >>> - Add ability to log commands executed in mod_xml_rpc >>> - Fix error when resuming a call on hold ? http://ift.tt/1rUJmMJ >>> - Fix missing Notify message record-route tag ? http://ift.tt/1ocyRZS >>> - Add new hard_mute control to allow apps to request low level mute >>> e.g. from the rtp stack level. Its used in mod_conference to avoid reading >>> audio while muted and possibly reduce some transcoding load >>> - Improve SIP OPTIONS ping generation by distributing them across an >>> interval ? http://ift.tt/1rUJlsi >>> - variable digits_dialed_filter to set regular expressions with () >>> captures and anything matched will be replaced with X?s in the CDR (useful >>> for filtering things like CreditCard numbers) ? http://ift.tt/1ocyS00 >>> - fix leak of nua handle due to reference counting. Effects all >>> calls with auth/challenge on INVITE >>> - as well as several other minor tweaks and changes >>> >>> There are also several other minor changes and bug fixes in this release! >>> >>> Along with this release we had to fix a small error in the main >>> FreeSWITCH GIT repository on the v1.4 branch. On your next pull of the >>> branch you may see a merge error. To resolve this error from your local >>> copy of the git repo while on the v1.4 branch issue the following command: >>> git reset --hard origin/v1.4 >>> Please note, if you have local patches you will want to rebase from the >>> main FreeSWITCH git repository to resolve the issue. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/7c99b724/attachment-0001.html From krice at freeswitch.org Tue Nov 4 16:34:17 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 4 Nov 2014 07:34:17 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> Message-ID: its was an opps in the process that didnt clean out the tree from the last build... we'll get that corrected shortly Ken Sent from my iPad > On Nov 4, 2014, at 02:24, Markus von Arx wrote: > > Hi Ken > > Thanks for the detailed release announcement! > > Any information about why the downloads are so huge for the 1.4.13 release? freeswitch-1.4.12.tar.bz2 was 37MB, freeswitch-1.4.13.tar.bz2 now is 442MB. The tar file contains 263MB of "rpmbuild", 168MB of "src_dist" and 126MB of "libs". > > Markus > > > 2014-11-04 0:42 GMT+01:00 Ken Rice : >> New Post on freeswitch.org from krice387 >> check it out at http://ift.tt/1rUJmfP >> FreeSWITCH 1.4.13 Released!! >> The FreeSWITCH Team is happy to announce the release of FreeSWITCH 1.4.13!!!! >> >> Tarball is available at http://ift.tt/1rUJls8 as well as packages for Debian and Centos6 in their respective repositories! >> >> This Release includes: >> >> Fix SegFault in switch_core_media.c ? http://ift.tt/1ocyP4k >> Updates for Xcode 6 on OSX Mavericks and Yosemite ? http://ift.tt/1rUJlsa >> Allow sub millisecond resolution for option ping times ? http://ift.tt/1rUJlsc >> Add ability to log commands executed in mod_xml_rpc >> Fix error when resuming a call on hold ? http://ift.tt/1rUJmMJ >> Fix missing Notify message record-route tag ? http://ift.tt/1ocyRZS >> Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load >> Improve SIP OPTIONS ping generation by distributing them across an interval ? http://ift.tt/1rUJlsi >> variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X?s in the CDR (useful for filtering things like CreditCard numbers) ? http://ift.tt/1ocyS00 >> fix leak of nua handle due to reference counting. Effects all calls with auth/challenge on INVITE >> as well as several other minor tweaks and changes >> There are also several other minor changes and bug fixes in this release! >> >> Along with this release we had to fix a small error in the main FreeSWITCH GIT repository on the v1.4 branch. On your next pull of the branch you may see a merge error. To resolve this error from your local copy of the git repo while on the v1.4 branch issue the following command: >> git reset --hard origin/v1.4 >> Please note, if you have local patches you will want to rebase from the main FreeSWITCH git repository to resolve the issue. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/ea1c22a9/attachment.html From grcamauer at gmail.com Tue Nov 4 17:47:42 2014 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 4 Nov 2014 11:47:42 -0300 Subject: [Freeswitch-users] Embedded Freeswitch In-Reply-To: References: <005c01cfd5bf$ce3c93a0$6ab5bae0$@comcast.net> Message-ID: Thanks Moy! Guillermo On Tue, Nov 4, 2014 at 5:16 AM, Moises Silva wrote: > Hi Guillermo, > > On Mon, Sep 22, 2014 at 12:55 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> Andrew, >> >> Thanks for the response. I am wondering if there are any examples using >> C/C++. >> > > You can have a look at the fscomm/ directory in FreeSWITCH git repo, then > peak at fshost.cpp ( > https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/fscomm/fshost.cpp > ) > > Pay particular attention to the FSHost::run() method calling the following > functions: > > switch_core_init() > switch_core_init_and_modload() > switch_core_runtime_loop() > > That's the gist of it to get started. In my opinion most of the time is > better to launch a lean freeswitch process (with only the modules you need > in your app) in the background and control it using ESL > > Cheers > > Moy > > Moises Silva > > Manager, Software Engineering > > > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, Ontario, Canada L3R 9R6 > > T +1 905 474 1990 x128 | toll-free in North America +1 800 388-2475 | F +1 > 905 474 9223 > > www.sangoma.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/9c3e8952/attachment.html From jmesquita at freeswitch.org Tue Nov 4 18:07:22 2014 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 4 Nov 2014 12:07:22 -0300 Subject: [Freeswitch-users] Embedded Freeswitch In-Reply-To: References: <005c01cfd5bf$ce3c93a0$6ab5bae0$@comcast.net> Message-ID: Guillermo, Even though the project has been inactive for a long time since I have no time to work on it, if you have questions I can try and help answer them. Sent from my iPhone > On Nov 4, 2014, at 11:47 AM, Guillermo Ruiz Camauer wrote: > > Thanks Moy! > > Guillermo > >> On Tue, Nov 4, 2014 at 5:16 AM, Moises Silva wrote: >> Hi Guillermo, >> >>> On Mon, Sep 22, 2014 at 12:55 PM, Guillermo Ruiz Camauer wrote: >>> Andrew, >>> >>> Thanks for the response. I am wondering if there are any examples using C/C++. >> >> You can have a look at the fscomm/ directory in FreeSWITCH git repo, then peak at fshost.cpp (https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/fscomm/fshost.cpp) >> >> Pay particular attention to the FSHost::run() method calling the following functions: >> >> switch_core_init() >> switch_core_init_and_modload() >> switch_core_runtime_loop() >> >> That's the gist of it to get started. In my opinion most of the time is better to launch a lean freeswitch process (with only the modules you need in your app) in the background and control it using ESL >> >> Cheers >> >> Moy >> >> Moises Silva >> Manager, Software Engineering >> >> Sangoma Technologies >> 100 Renfrew Drive, Suite 100, Markham, Ontario, Canada L3R 9R6 >> T +1 905 474 1990 x128 | toll-free in North America +1 800 388-2475 | F +1 905 474 9223 >> www.sangoma.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/a40e7fbc/attachment-0001.html From matsumoto at itsherpa.com Tue Nov 4 18:51:54 2014 From: matsumoto at itsherpa.com (=?UTF-8?B?5p2+5pys56WQ5b+X?=) Date: Wed, 5 Nov 2014 00:51:54 +0900 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: Dear Stanislav Sinyagin Thank you for your help. ????????????????????????????? ???Perl??? ?????????????????????????wav?????????????????? ????????????????????? my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; ???$ con-> bgapi ($ api_cmd); ???????????? fs_cli ? ?conference ************* play? ????????????????????????? ????????????? ??????????? On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin wrote: > Dear Matsumoto-san, > > I think it will be easier if you write in Japanese, then it will be clear > how we could help. I know a few Japanese-speaking colleagues who may help > in communicating. > > > > > On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > >> Hello >> >> I have two issues. >> >> I am writing in Perl. >> >> While 2 people are talking in a conference room, the one person want to >> play the sound. >> >> In "Caller-Username", can you get useless. >> I have tried the above but, Member: it will not become a *** not found.. >> >> my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader >> ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; >> >> >> I want to play the above in the background >> It can not play in the next program. >> my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader >> ("Conference-Name"), /etc/a.wav); >> >> $ con-> bgapi ($ api_cmd); >> >> Best regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/d208e5e3/attachment.html From anthony.minessale at gmail.com Tue Nov 4 19:39:16 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 4 Nov 2014 10:39:16 -0600 Subject: [Freeswitch-users] mod_verto unwanted profile shutdown In-Reply-To: References: <1853D18C-7264-45FF-9B17-29487807EB48@jerris.com> Message-ID: What did it say in the ... That is probably where it told you the problem. Try running in debug 10 (set in verto.conf.xml at the top) and console loglevel debug and what OS is it? Report all of the findings and the logs to jira http://jira.freeswitch.org The mailing list is not a place for issue reports. On Tue, Nov 4, 2014 at 3:52 AM, S?ndor Bal?zs < balazs.sandor at virtual-call-center.hu> wrote: > It not worked. Same issue > > 2014-11-04 10:18:09.731503 [CONSOLE] sofia_presence.c:1618 Event Thread > Started > freeswitch at default> verto status > Name Type Data > State > > ================================================================================================= > mine profile ws: > 192.168.1.33:8081 RUNNING > mine profile wss: > 192.168.1.33:8082 RUNNING > > ================================================================================================= > 1 profile , 0 clients > > ... > 2014-11-04 10:23:00.051524 [INFO] mod_verto.c:4524 profile mine shutdown, > Waiting for 0 threads > 2014-11-04 10:23:00.051524 [INFO] mod_verto.c:4535 mine Thread ending > > > ?dv?zlettel: > > S?ndor Bal?zs > > Szoftver fejleszt? > > > > Virtual Call Center > > MUNICH | BUDAPEST > | WARSAW > > > Telefon: +36 1 999 7400 > > Web: www.virtual-call-center.hu > > > > > 2014-11-03 14:40 GMT+01:00 Michael Jerris : > >> Does it do the same thing if you bind to a real ip instead of 0.0.0.0? >> >> On Nov 3, 2014, at 4:25 AM, Bal?zs S?ndor < >> balazs.sandor at virtual-call-center.eu> wrote: >> >> Hi all. >> >> I'm testing mod_verto. >> After a while (5-10 minutes) it always disconnecting. >> I do nothing with FS (its sleeping) >> >> It seems to be normal shutdown >> 2014-11-03 08:50:30.495346 [INFO] mod_verto.c:4130 Secure key and cert >> specified >> 2014-11-03 08:50:30.495370 [INFO] mod_verto.c:4218 mine Bound to >> 0.0.0.0:8081 >> 2014-11-03 08:50:30.495377 [INFO] mod_verto.c:4218 mine Bound to >> 0.0.0.0:8082 >> -- >> 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4524 profile mine shutdown, >> Waiting for 0 threads >> 2014-11-03 09:00:29.211502 [INFO] mod_verto.c:4535 mine Thread ending >> >> I attached a detailed log >> >> Best regards, >> >> Bal?zs S?NDOR >> Software Developer >> >> >> >> Virtual Call Center >> MUNICH | BUDAPEST >> | WARSAW >> >> Phone: +44 (0) 863 801 69 >> Web: www.virtual-call-center.eu >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/8fc571d6/attachment-0001.html From joelewhite at gmail.com Tue Nov 4 21:03:56 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 4 Nov 2014 13:03:56 -0500 Subject: [Freeswitch-users] PostgreSQL - p_key In-Reply-To: References: <543F3C95.5050807@quentustech.com> Message-ID: Well here is what I have done, I think that it should work. The PK is a SERIAL DataType and I went into the sequence properties and set it to CYCLED. The max value is 9223372036854775807 and will re-cycle once that number is reached. I only added this to the channels table right now. I have not seen any issues yet. On Mon, Nov 3, 2014 at 4:50 PM, Michael Jerris wrote: > I've used bigint with a default of a sequence in the past to good results. > make sure to add the option to not recreate the tables so we don't mess up > your fields when we change scheme and when upgrading review for schema > changes and make them manually > > > On Monday, November 3, 2014, Joel White wrote: > >> Is there another datatype that would be more suitable. I plan to have >> massive amounts of traffic. Would the time datatype be more suitable? >> >> >> >> On Mon, Oct 20, 2014 at 12:26 PM, Joel White >> wrote: >> >>> Thank you Chris >>> >>> On Fri, Oct 17, 2014 at 2:08 PM, Chris Tunbridge >>> wrote: >>> >>>> Serial is identical to mysql's Auto Increment, so i'm going to say yes. >>>> >>>> On Fri, Oct 17, 2014 at 9:36 AM, Joel White >>>> wrote: >>>> >>>>> Chris and William, >>>>> >>>>> Can those added PKs have a data type of SERIAL? >>>>> >>>>> >>>>> >>>>> On Wed, Oct 15, 2014 at 11:33 PM, William King < >>>>> william.king at quentustech.com> wrote: >>>>> >>>>>> Yes, you can add columns to those tables with no issues. Try it and >>>>>> see. >>>>>> >>>>>> William King >>>>>> Senior Engineer >>>>>> Quentus Technologies, INC >>>>>> 1037 NE 65th St Suite 273 >>>>>> Seattle, WA 98115 >>>>>> Main: (877) 211-9337 >>>>>> Office: (206) 388-4772 >>>>>> Cell: (253) 686-5518william.king at quentustech.com >>>>>> >>>>>> On 10/8/14 9:44 AM, Joel White wrote: >>>>>> >>>>>> Is it feasible to add Primary Keys to all FreeSWITCH tables or >>>>>> would this break some functionality? >>>>>> >>>>>> I am asking because I am working on a multi-master replication setup >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/64ac6f7b/attachment.html From krice at freeswitch.org Tue Nov 4 21:41:47 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 04 Nov 2014 18:41:47 +0000 Subject: [Freeswitch-users] Freeswitch Week in Revew (Master Branch) September 28th-October 5th Message-ID: <54591debdffcd_2821e7f33834772@ip-10-69-135-107.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1E2VPVl Freeswitch Week in Revew (Master Branch) September 28th-October 5th Hello, again. This week in the FreeSWITCH master branch we had 48 commits. We had a few new features added, like, tab completion for api commands in mod_gsmopen, a new feature to filter the SDP on bypass_media calls to remove or limit codecs, and a way to globally disable system commands by setting global var disable_system_api_commands=true. The following bugs were squashed: d619017 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1uqVaxG;] da51603 FS-6757 FS-6713 FS-6868 FS-6863 FS-6858 #resolve #comment 5 bugs one typo. From commit 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: http://ift.tt/1sh2sks] fbe857e FS-6644 Fix ptime from known broken endpoints on re-invite [Jira: http://ift.tt/1E2VGkM] 0150c86 FS-6854 Patch for SIP lockout on WebRTC in mod_verto [Jira: http://ift.tt/1uqV8WN] 644b41f FS-6874 Fix for multipart invite format in mod_sofia [Jira: http://ift.tt/1uqV9d2] 35aeae0 FS-6822 #comment The code in question appears to have been added by me (18f20e24).? I think this patch is the correct solution. [Jira: http://ift.tt/1uqV9d6] 8db31f9 Fix some recovery issues with dynamic payloads ae5d865 FS-6884 #comment Fix for simple warnings in mod_sofia and mod_conference [Jira: http://ift.tt/1E2VGBa] 01bf422 FS-6888 Fix regression from refactoring new feature [Jira: http://ift.tt/1uqVaxL] bde2e2d FS-6889 Fix for Reinvite Codec Error when using bypass_media_resume_on_hold and bypass_media_after_hold in mod_sofia [Jira: http://ift.tt/1E2VIsW] b2ae5f4 Fixed a few bugs in the new features in mod_sofia afd6875 FS-6781 Adding a check to ensure consistency in mod_valet_parking [Jira: http://ift.tt/1uqVaxP] New features that were added: a94fbe8 Add tab completion for api commands in mod_gsmopen 92a66fb Improve adaptive jitter buffer ascending check in switch_stfu 24084ad Add new feature to filter the SDP on bypass_media calls to remove or limit codecs. 8e408e9 FS-6865 #resolve add XMPP priority to dingaling [Jira: http://ift.tt/1E2VIJi] 789e148 FS-6880 Improvements to filtering invites in the core. [Jira: http://ift.tt/1uqVaxR] b3d7191 FS-6870 vs2010 and vs2012 fix [Jira: http://ift.tt/1E2VGBl] 91ffe17 Use OPUS_APPLICATION_VOIP always to get FEC and filtering in mod_opus 9e91753 FS-6886 Addition rtp_disable_hold when bypass_media is active and addition of ignoring unhold in mod_sofia [Jira: http://ift.tt/1uqVaxT] 6bfc05b FS-6887 Add always_auto_adjust parameter [Jira: http://ift.tt/1E2VGRD] 43733a6 FS-6886 #comment addition of ignoring unhold as well to mod_sofia [Jira: http://ift.tt/1uqVaxT] 10a3fa5 Add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis 0d1f5d0 Add way to globally disable system commands by setting global var disable_system_api_commands=true Improvements in cross platform build supports: 812f911 Fix reference to libressl in makefile on OpenBSD a39db86 FS-6870 #vs2012 and vs2010 make download of openssl dependent [Jira: http://ift.tt/1E2VGBl] d17f14e Make sure to pass along appropriate configure flags to sub-configure?s when cross compiling d52cb33 Fix trivial vs2010 build errors The complete list of commits can be found here: 2014_9_28-2014_10_5 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/23e39d85/attachment.html From sdevoy at bizfocused.com Tue Nov 4 22:16:10 2014 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 4 Nov 2014 19:16:10 +0000 Subject: [Freeswitch-users] XML DialPlan help In-Reply-To: <00000145c37e4390-711ac293-b21b-4444-ae7d-0915baae64a3-000000@email.amazonses.com> References: <0281dd277f51451ea35c584e938ec3f5@BN1PR01MB246.prod.exchangelabs.com> <00000145c37e4390-711ac293-b21b-4444-ae7d-0915baae64a3-000000@email.amazonses.com> Message-ID: Yes, I just added country code 31. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, May 03, 2014 3:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] XML DialPlan help You can use tonestream to play a dialtone (probably), but this will be live during a SIP call to the FS server, not on the phone. You have to control when it times out to start the call-- on my calling card I have a timeout or allow the user to press # to indicate that they are done. -Avi On Sat, May 3, 2014 at 6:10 PM, Sean Devoy > wrote: OK, I saw that piece. But what about giving them another dialtone? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Saturday, May 03, 2014 3:37 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] XML DialPlan help http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits On Saturday, May 3, 2014, Sean Devoy > wrote: I swear I have read this, but now that I need it I can?t find it. My user wants to dial international. I want to require them to do SOME extra step other than dialing 0113010? to call the Netherlands. Oddly enough, it is cheaper than US termination ? go figure. Ideally, I would like them dial some extension like 777, hear a new dialtone and then accept the international number. I might even like to play a message that says ?All charges for International are their responsibility and not part of their current package? before the dialtone. Ideas? Thanks again, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141104/27450f69/attachment-0001.html From jerry.allen at vistabeam.com Tue Nov 4 22:36:10 2014 From: jerry.allen at vistabeam.com (Jerry Allen) Date: Tue, 4 Nov 2014 12:36:10 -0700 Subject: [Freeswitch-users] Vulnerability Testing Message-ID: <121c01cff866$98028b90$c807a2b0$@vistabeam.com> I believe I have my system "ready for use" Are there any exploit testing tools that test for the ability to make calls etc. off of a system? I have tested with sipvicious and sipsak but I am not positive that I am testing correctly. I would like a real hack test. The system has been "wide open" for a few weeks, fine-tuned fail2ban as well as the addition of a firewall that protects my FS machine (hopefully) and there have been no outbound calls made that I did not make there have been hundreds of various hackers/scanners that fail2ban blocked etc. I am not exactly sure where to go to make sure it is not a big bomb waiting to be set off. Jerry Allen From william.king at quentustech.com Tue Nov 4 22:48:32 2014 From: william.king at quentustech.com (William King) Date: Tue, 04 Nov 2014 11:48:32 -0800 Subject: [Freeswitch-users] Vulnerability Testing In-Reply-To: <121c01cff866$98028b90$c807a2b0$@vistabeam.com> References: <121c01cff866$98028b90$c807a2b0$@vistabeam.com> Message-ID: <54592D90.9070806@quentustech.com> I would suggest contacting consulting at freeswitch.org and asking for a security review. It'll be more effective than just asking for black box scanning on the users list. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 11/04/2014 11:36 AM, Jerry Allen wrote: > I believe I have my system "ready for use" Are there any exploit testing tools that test for the ability to make calls etc. off of a system? I have tested with sipvicious and sipsak but I am not positive that I am testing correctly. > I would like a real hack test. The system has been "wide open" for a few weeks, fine-tuned fail2ban as well as the addition of a firewall that protects my FS machine (hopefully) and there have been no outbound calls made that I did not make there have been hundreds of various hackers/scanners that fail2ban blocked etc. > > I am not exactly sure where to go to make sure it is not a big bomb waiting to be set off. > > Jerry Allen > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jerry.allen at vistabeam.com Tue Nov 4 23:01:44 2014 From: jerry.allen at vistabeam.com (Jerry Allen) Date: Tue, 4 Nov 2014 13:01:44 -0700 Subject: [Freeswitch-users] Vulnerability Testing In-Reply-To: <54592D90.9070806@quentustech.com> References: <121c01cff866$98028b90$c807a2b0$@vistabeam.com> <54592D90.9070806@quentustech.com> Message-ID: <122301cff86a$298f1b20$7cad5160$@vistabeam.com> Thank you for that idea. I just did so.... Jerry Allen -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William King Sent: Tuesday, November 04, 2014 12:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Vulnerability Testing I would suggest contacting consulting at freeswitch.org and asking for a security review. It'll be more effective than just asking for black box scanning on the users list. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 11/04/2014 11:36 AM, Jerry Allen wrote: > I believe I have my system "ready for use" Are there any exploit testing tools that test for the ability to make calls etc. off of a system? I have tested with sipvicious and sipsak but I am not positive that I am testing correctly. > I would like a real hack test. The system has been "wide open" for a few weeks, fine-tuned fail2ban as well as the addition of a firewall that protects my FS machine (hopefully) and there have been no outbound calls made that I did not make there have been hundreds of various hackers/scanners that fail2ban blocked etc. > > I am not exactly sure where to go to make sure it is not a big bomb waiting to be set off. > > Jerry Allen > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From tahir at ictinnovations.com Tue Nov 4 23:40:52 2014 From: tahir at ictinnovations.com (Tahir Almas) Date: Wed, 5 Nov 2014 01:40:52 +0500 Subject: [Freeswitch-users] ICTFax Version 3.0 released Message-ID: Please to announce the release of ICTFax Version 3.0 , for more information please visit http://www.ictinnovations.com/content/ictfax-version-30-fax-over-ip-solution-released-now http://www.icfax.org regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/94b94c0e/attachment.html From jerry.allen at vistabeam.com Wed Nov 5 02:13:44 2014 From: jerry.allen at vistabeam.com (Jerry Allen) Date: Tue, 4 Nov 2014 16:13:44 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> Message-ID: <124801cff884$fc5368d0$f4fa3a70$@vistabeam.com> Does 1.4.13 still have issues with mod_com_G729? Last I tried and also heard on this list there were issues. Jerry Allen From krice at freeswitch.org Wed Nov 5 03:09:32 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 05 Nov 2014 00:09:32 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review(Master Branch) October 5th- 12th Message-ID: <54596abc45b9_cc4b78133868018@ip-10-156-243-37.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1AdJhhd Freeswitch Week in Review(Master Branch) October 5th- 12th Hello, again. This week in the FreeSWITCH master branch we had 28 commits. Some of the feature work done this week included improvements to the jitter buffer, the addition of uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message in mod_commands, and a 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly in mod_sofia. New features that were added: da43bde Add some calculations to jitter buffer related to judging the optimal size eaaf946 FS-6897: uuid_send_info enhancement that allows setting the Content-Type of the SIP INFO message in mod_commands [Jira: http://ift.tt/1txlhzp] a4f840b More jitter buffer improvements 855cc4b Add 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly in mod_sofia The following bugs were squashed: 397ec5a Fixed a jitter buffer logic error 490efb7 FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration in mod_sofia [Jira: http://ift.tt/1AdJhhm] 7ec7c92 OPENZAP-220 fix blocked into read and add cause for a correct hangup 4a5e36d Fixed switch_pgsql_next_result_timed() to use switch_micro_time_now() to be consistent 2051a86 FS-6889 Fix for Reinvite Codec Error when using bypass_media_resume_on_hold and bypass_media_after_hold in mod_sofia [Jira: http://ift.tt/1E2VIsW] d48057e FS-6890 Fix to account for lost frames during ptime detection 28bc992 Fix error in SRGS grammar parser in mod_rayo 43c2c6d FS-6815 Fix for fs_encode issue [Jira: http://ift.tt/1txlkv7] 66dafbd FS-6902 Add patch to make native SILK encoded playback problem obvious and fail on record and playback [Jira: http://ift.tt/1txlhzr] e4e9b1b Fix to have resume media on hold not send invite back out at the holder but rather enable media in the 200ok in mod_sofia In terms of stability these were the use cases that were fixed: 2514de9 Fix obvious seg in setting a record file name to every participant and not checking for the recording member which does not have a session in mod_conference b5294c5 Fix crash on transport=tls with non-TLS profile in mod_sofia Improvements in cross platform build supports: 6146efd FS-6870 Fix for build issue with vs2010 [Jira: http://ift.tt/1E2VGBl] The complete list of commits can be found here:2014_10_5-2014_10_12 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/6c031c75/attachment.html From dujinfang at gmail.com Wed Nov 5 04:52:10 2014 From: dujinfang at gmail.com (Seven Du) Date: Wed, 5 Nov 2014 09:52:10 +0800 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: I don?t quite understand you but you should not put SPACE between % and s. Try to paste the real code and debug logs on pastebin to get better help. On Tuesday, November 4, 2014 at 11:51 PM, ???? wrote: > Dear Stanislav Sinyagin > > Thank you for your help. > > ????????????????????????????? > > ???Perl??? > > ?????????????????????????wav?????????????????? > > ????????????????????? > my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > ???$ con-> bgapi ($ api_cmd); ????????????? > > fs_cli ? ?conference ************* play? ????????????????????????? > > ????????????? > > ??????????? > > > > On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin wrote: > > Dear Matsumoto-san, > > > > I think it will be easier if you write in Japanese, then it will be clear how we could help. I know a few Japanese-speaking colleagues who may help in communicating. > > > > > > > > > > On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > > > Hello > > > > > > I have two issues. > > > > > > I am writing in Perl. > > > > > > While 2 people are talking in a conference room, the one person want to play the sound. > > > > > > In "Caller-Username", can you get useless. > > > I have tried the above but, Member: it will not become a *** not found.. > > > > > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > > > > > > > I want to play the above in the background > > > It can not play in the next program. > > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav); > > > > > > $ con-> bgapi ($ api_cmd); > > > > > > Best regards > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/ad690cb1/attachment-0001.html From ssinyagin at gmail.com Wed Nov 5 05:00:02 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 5 Nov 2014 03:00:02 +0100 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: you need to make sure that those headers ("Conference-Name", "Caller-Username") are really present in the event. probably it's just the wrong event which is not related to a conference. You can print the contents of an event to a standard output and see if the headers that you need are present in it: print $e->serialize; Also it makes sense to print the $api_cmd variable, and see if its value is what you expect. Also, sometimes the WAV file is simply not there :) On Tue, Nov 4, 2014 at 4:51 PM, ???? wrote: > Dear Stanislav Sinyagin > > Thank you for your help. > > ????????????????????????????? > > ???Perl??? > > ?????????????????????????wav?????????????????? > > ????????????????????? > my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > ???$ con-> bgapi ($ api_cmd); ????????????? > > fs_cli ? ?conference ************* play? ????????????????????????? > > ????????????? > > ??????????? > > > > On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin > wrote: >> >> Dear Matsumoto-san, >> >> I think it will be easier if you write in Japanese, then it will be clear >> how we could help. I know a few Japanese-speaking colleagues who may help in >> communicating. >> >> >> >> >> On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: >>> >>> Hello >>> >>> I have two issues. >>> >>> I am writing in Perl. >>> >>> While 2 people are talking in a conference room, the one person want to >>> play the sound. >>> >>> In "Caller-Username", can you get useless. >>> I have tried the above but, Member: it will not become a *** not found.. >>> >>> my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader >>> ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; >>> >>> >>> I want to play the above in the background >>> It can not play in the next program. >>> my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader >>> ("Conference-Name"), /etc/a.wav); >>> >>> $ con-> bgapi ($ api_cmd); >>> >>> Best regards >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdelawarde at wirelessmundi.com Wed Nov 5 12:01:04 2014 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 05 Nov 2014 10:01:04 +0100 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: <124801cff884$fc5368d0$f4fa3a70$@vistabeam.com> References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> <124801cff884$fc5368d0$f4fa3a70$@vistabeam.com> Message-ID: <1415178064.8557.55.camel@luna.madrid.commsmundi.com> Not anymore you're not! Just make sure you use the latest version from http://files.freeswitch.org/g729/ Fran?ois. On Tue, 2014-11-04 at 16:13 -0700, Jerry Allen wrote: > Does 1.4.13 still have issues with mod_com_G729? Last I tried and also heard on this list there were issues. > > Jerry Allen > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/f9bf0c5a/attachment.html From GB at cm.nl Wed Nov 5 12:43:03 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 5 Nov 2014 10:43:03 +0100 Subject: [Freeswitch-users] ESL: Which event is generated when an api command completed? Message-ID: Hello, When using Outbound ESL in async full mode and executing a API command like uuid_transfer or uuid_bridge using bgapi, which channel event is generated to notify the command execution completed? For dialplan applications it is the CHANNEL_EXECUTE_COMPLETED event. That I know. All I got is this: Command Executed: uuid_transfer 563c9750-64cc-11e4-8cd8-4fc1451152d2 -both park_extension XML park_ext Bgapi returned an ESLevent: { "Event-Name": "SOCKET_DATA", "Content-Type": "command/reply", "Reply-Text": "+OK Job-UUID: ede6c869-0bfa-4e07-8490-04973b6ba971", "Job-UUID": "ede6c869-0bfa-4e07-8490-04973b6ba971" } However, a channel event named SOCKET_DATA did not show up when listening for channel events using the RecvEvent method. What I'm trying to do is use bgapi or sendMSG to execute an application/command, and use the RecvEvent method to wait for a channel event that indicates the application/command completed execution. So if I were to use the playback application, I'd send it using the SendMSG method, and use RecvEvent to wait for the CHANNEL_EXECUTE_COMPLETED event containing the UUID set for this playback. This works fine for dialplan applications but I don't know which event to listen for when using api commands. I hope someone could point me into the right direction. Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/a38b9dd3/attachment.html From steveayre at gmail.com Wed Nov 5 12:48:38 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 5 Nov 2014 09:48:38 +0000 Subject: [Freeswitch-users] ESL: Which event is generated when an api command completed? In-Reply-To: References: Message-ID: For bgapi it's https://wiki.freeswitch.org/wiki/Event_list#BACKGROUND_JOB On 5 November 2014 09:43, Grant Bagdasarian wrote: > Hello, > > > > When using Outbound ESL in async full mode and executing a API command > like uuid_transfer or uuid_bridge using bgapi, which channel event is > generated to notify the command execution completed? > > For dialplan applications it is the CHANNEL_EXECUTE_COMPLETED event. That > I know. > > > > All I got is this: > > > > Command Executed: uuid_transfer 563c9750-64cc-11e4-8cd8-4fc1451152d2 -both > park_extension XML park_ext > > Bgapi returned an ESLevent: > > { > > "Event-Name": "SOCKET_DATA", > > "Content-Type": "command/reply", > > "Reply-Text": "+OK Job-UUID: > ede6c869-0bfa-4e07-8490-04973b6ba971", > > "Job-UUID": "ede6c869-0bfa-4e07-8490-04973b6ba971" > > } > > > > However, a channel event named SOCKET_DATA did not show up when listening > for channel events using the RecvEvent method. > > > > What I?m trying to do is use bgapi or sendMSG to execute an > application/command, and use the RecvEvent method to wait for a channel > event that indicates the application/command completed execution. > > So if I were to use the playback application, I?d send it using the > SendMSG method, and use RecvEvent to wait for the CHANNEL_EXECUTE_COMPLETED > event containing the UUID set for this playback. > > This works fine for dialplan applications but I don?t know which event to > listen for when using api commands. > > > > I hope someone could point me into the right direction. > > > > Regards, > > > Grant > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/d8147fcf/attachment.html From wilddragon at sina.com Wed Nov 5 10:15:46 2014 From: wilddragon at sina.com (wilddragon at sina.com) Date: Wed, 5 Nov 2014 15:15:46 +0800 Subject: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge References: <54596abc45b9_cc4b78133868018@ip-10-156-243-37.mail> Message-ID: <2014110515153895656810@sina.com> I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. who can help me? thanks. wilddragon at sina.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/b82a9eeb/attachment-0001.html From jerry.allen at vistabeam.com Wed Nov 5 17:23:15 2014 From: jerry.allen at vistabeam.com (jerry.allen at vistabeam.com) Date: Wed, 5 Nov 2014 07:23:15 -0700 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: <1415178064.8557.55.camel@luna.madrid.commsmundi.com> References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> <124801cff884$fc5368d0$f4fa3a70$@vistabeam.com> <1415178064.8557.55.camel@luna.madrid.commsmundi.com> Message-ID: <379e570c73a7e9a4ceffb3e0b86b224e.squirrel@mail.vistabeam.com> I must be using the wrong installer or something. Installs fine, licenses stay in place but g729_info results in a full Freeswitch crash. Jerry Allen > Not anymore you're not! Just make sure you use the latest version from > http://files.freeswitch.org/g729/ > > Fran??ois. > > > On Tue, 2014-11-04 at 16:13 -0700, Jerry Allen wrote: > >> Does 1.4.13 still have issues with mod_com_G729? Last I tried and also >> heard on this list there were issues. >> >> Jerry Allen >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dev.2981988 at gmail.com Wed Nov 5 17:33:03 2014 From: dev.2981988 at gmail.com (divyesh kamothi) Date: Wed, 5 Nov 2014 20:03:03 +0530 Subject: [Freeswitch-users] srtp-sdp header tag. Message-ID: Hello team I updated freeswitch to 1.2 to 1.5. now here issue is when in using srtp. with freeswitch 1.2 :: In SDP value after a=crypto:0 <= this is 0 with freeswitch 1.5 :: In SDP value after a=crypto:8 <= this is 8 is there any configuration or other way using which i can change digit to 0. -- Thanks and Regards Divyesh Kamothi PH : - 00-91-84605-18040 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/e20f3df4/attachment.html From brian at freeswitch.org Wed Nov 5 18:33:26 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Nov 2014 09:33:26 -0600 Subject: [Freeswitch-users] srtp-sdp header tag. In-Reply-To: References: Message-ID: It doesn't have to be zero. Why are you thinking it MUST be zero? Its an index. On Wed, Nov 5, 2014 at 8:33 AM, divyesh kamothi wrote: > Hello team > > > I updated freeswitch to 1.2 to 1.5. now here issue is when in using srtp. > > with freeswitch 1.2 :: > In SDP value after a=crypto:0 <= this is 0 > > with freeswitch 1.5 :: > In SDP value after a=crypto:8 <= this is 8 > > is there any configuration or other way using which i can change digit to > 0. > > > > > -- > Thanks and Regards > Divyesh Kamothi > PH : - 00-91-84605-18040 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/232fe9cd/attachment.html From brian at freeswitch.org Wed Nov 5 18:35:19 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Nov 2014 09:35:19 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.4.13 Released!! In-Reply-To: <379e570c73a7e9a4ceffb3e0b86b224e.squirrel@mail.vistabeam.com> References: <545812cfdb9d4_966b83328616ce@ip-10-156-212-88.mail> <124801cff884$fc5368d0$f4fa3a70$@vistabeam.com> <1415178064.8557.55.camel@luna.madrid.commsmundi.com> <379e570c73a7e9a4ceffb3e0b86b224e.squirrel@mail.vistabeam.com> Message-ID: Please collect the backtrace, the only crash associated with calling g729_info was FS-6911 which you may be hitting if you're not on 1.4.13. If the issue persists please file a JIRA, the mailing list is not the proper place to post bug reports. On Wed, Nov 5, 2014 at 8:23 AM, wrote: > I must be using the wrong installer or something. Installs fine, licenses > stay in place but g729_info results in a full Freeswitch crash. > > Jerry Allen > > > Not anymore you're not! Just make sure you use the latest version from > > http://files.freeswitch.org/g729/ > > > > Fran??ois. > > > > > > On Tue, 2014-11-04 at 16:13 -0700, Jerry Allen wrote: > > > >> Does 1.4.13 still have issues with mod_com_G729? Last I tried and also > >> heard on this list there were issues. > >> > >> Jerry Allen > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/7f78cb8f/attachment.html From spencer at whiteskycommunications.com Wed Nov 5 20:18:41 2014 From: spencer at whiteskycommunications.com (Spencer Thomason) Date: Wed, 5 Nov 2014 11:18:41 -0600 Subject: [Freeswitch-users] Gateway Status 908 Message-ID: Hello, I?m a bit confused by gateway timeout status 908. On a REGISTER timeout, the status is set to 908, gateway_ptr->failure_status = 908; Is this a typo? Should it not be 408? I know the 908-retry-seconds parameter was added recently, I guess i?m just confused that its not 408-retry-seconds. Thanks, Spencer From brian at freeswitch.org Wed Nov 5 20:26:06 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Nov 2014 11:26:06 -0600 Subject: [Freeswitch-users] Gateway Status 908 In-Reply-To: References: Message-ID: https://freeswitch.org/jira/browse/FS-5751 This JIRA is about this exact topic, I think its up for debate, I would put your comments on that JIRA, I'll poke Travis to review it again. On Wed, Nov 5, 2014 at 11:18 AM, Spencer Thomason < spencer at whiteskycommunications.com> wrote: > Hello, > I?m a bit confused by gateway timeout status 908. On a REGISTER timeout, > the status is set to 908, gateway_ptr->failure_status = 908; Is this a > typo? Should it not be 408? I know the 908-retry-seconds parameter was > added recently, I guess i?m just confused that its not 408-retry-seconds. > > Thanks, > Spencer > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/9311c65a/attachment-0001.html From GB at cm.nl Wed Nov 5 23:03:21 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 5 Nov 2014 21:03:21 +0100 Subject: [Freeswitch-users] ESL: Which event is generated when an api command completed? In-Reply-To: References: , Message-ID: That's it! Thanks! ________________________________________ Van: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] namens Steven Ayre [steveayre at gmail.com] Verzonden: woensdag 5 november 2014 10:48 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] ESL: Which event is generated when an api command completed? For bgapi it's https://wiki.freeswitch.org/wiki/Event_list#BACKGROUND_JOB On 5 November 2014 09:43, Grant Bagdasarian > wrote: Hello, When using Outbound ESL in async full mode and executing a API command like uuid_transfer or uuid_bridge using bgapi, which channel event is generated to notify the command execution completed? For dialplan applications it is the CHANNEL_EXECUTE_COMPLETED event. That I know. All I got is this: Command Executed: uuid_transfer 563c9750-64cc-11e4-8cd8-4fc1451152d2 -both park_extension XML park_ext Bgapi returned an ESLevent: { "Event-Name": "SOCKET_DATA", "Content-Type": "command/reply", "Reply-Text": "+OK Job-UUID: ede6c869-0bfa-4e07-8490-04973b6ba971", "Job-UUID": "ede6c869-0bfa-4e07-8490-04973b6ba971" } However, a channel event named SOCKET_DATA did not show up when listening for channel events using the RecvEvent method. What I?m trying to do is use bgapi or sendMSG to execute an application/command, and use the RecvEvent method to wait for a channel event that indicates the application/command completed execution. So if I were to use the playback application, I?d send it using the SendMSG method, and use RecvEvent to wait for the CHANNEL_EXECUTE_COMPLETED event containing the UUID set for this playback. This works fine for dialplan applications but I don?t know which event to listen for when using api commands. I hope someone could point me into the right direction. Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From aqsyounas at gmail.com Wed Nov 5 23:30:06 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 6 Nov 2014 01:30:06 +0500 Subject: [Freeswitch-users] Does freeswitch support acc format? Message-ID: Hi, I am new to freeswitch. Does anybody know freeswitch support acc format? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/88afca8a/attachment.html From brian at freeswitch.org Thu Nov 6 00:31:09 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 5 Nov 2014 15:31:09 -0600 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: I think you mean AAC? On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: > Hi, > I am new to freeswitch. Does anybody know freeswitch support acc format? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/c1b9eaca/attachment.html From blasterjr at gmail.com Thu Nov 6 01:59:03 2014 From: blasterjr at gmail.com (Chris Tunbridge) Date: Wed, 5 Nov 2014 15:59:03 -0700 Subject: [Freeswitch-users] PostgreSQL - p_key In-Reply-To: References: <543F3C95.5050807@quentustech.com> Message-ID: if you had 100 years of calls, having say 4 channels per call, you could have 63,173,781,074,347 calls per day that's 63 trillion calls... i wouldn't worry too much about seeing repeat/cycle anytime soon. You are talking about having 9 quintillion records, it should be safe to say that with bigint/serial setup like that you will be fine... you're much more likely to have a database server crash than run out of indexes at that point. On Tue, Nov 4, 2014 at 11:03 AM, Joel White wrote: > Well here is what I have done, I think that it should work. The PK is a > SERIAL DataType and I went into the sequence properties and set it to > CYCLED. The max value is 9223372036854775807 and will re-cycle once that > number is reached. > > I only added this to the channels table right now. I have not seen any > issues yet. > > On Mon, Nov 3, 2014 at 4:50 PM, Michael Jerris wrote: > >> I've used bigint with a default of a sequence in the past to good >> results. make sure to add the option to not recreate the tables so we >> don't mess up your fields when we change scheme and when upgrading review >> for schema changes and make them manually >> >> >> On Monday, November 3, 2014, Joel White wrote: >> >>> Is there another datatype that would be more suitable. I plan to have >>> massive amounts of traffic. Would the time datatype be more suitable? >>> >>> >>> >>> On Mon, Oct 20, 2014 at 12:26 PM, Joel White >>> wrote: >>> >>>> Thank you Chris >>>> >>>> On Fri, Oct 17, 2014 at 2:08 PM, Chris Tunbridge >>>> wrote: >>>> >>>>> Serial is identical to mysql's Auto Increment, so i'm going to say yes. >>>>> >>>>> On Fri, Oct 17, 2014 at 9:36 AM, Joel White >>>>> wrote: >>>>> >>>>>> Chris and William, >>>>>> >>>>>> Can those added PKs have a data type of SERIAL? >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Oct 15, 2014 at 11:33 PM, William King < >>>>>> william.king at quentustech.com> wrote: >>>>>> >>>>>>> Yes, you can add columns to those tables with no issues. Try it >>>>>>> and see. >>>>>>> >>>>>>> William King >>>>>>> Senior Engineer >>>>>>> Quentus Technologies, INC >>>>>>> 1037 NE 65th St Suite 273 >>>>>>> Seattle, WA 98115 >>>>>>> Main: (877) 211-9337 >>>>>>> Office: (206) 388-4772 >>>>>>> Cell: (253) 686-5518william.king at quentustech.com >>>>>>> >>>>>>> On 10/8/14 9:44 AM, Joel White wrote: >>>>>>> >>>>>>> Is it feasible to add Primary Keys to all FreeSWITCH tables or >>>>>>> would this break some functionality? >>>>>>> >>>>>>> I am asking because I am working on a multi-master replication setup >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>>>>>> >>>>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141105/e746424b/attachment-0001.html From manish.talwar at nexxuspg.com Thu Nov 6 10:07:06 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Thu, 6 Nov 2014 07:07:06 +0000 Subject: [Freeswitch-users] SIP trunking with Nexmo Message-ID: <1415306341755.98271@nexxuspg.com> Hi,? I have? make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external?" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name ?Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: 919818753995 here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED? Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) ?as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/f8ab35d3/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: FsCall.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/f8ab35d3/attachment-0001.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: Nexmo.xml Type: text/xml Size: 2603 bytes Desc: Nexmo.xml Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/f8ab35d3/attachment-0001.xml From wilddragon at 163.com Thu Nov 6 10:12:09 2014 From: wilddragon at 163.com (Daniel Kou) Date: Thu, 6 Nov 2014 15:12:09 +0800 Subject: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge Message-ID: <02fb01cff990$fb89c270$f29d4750$@163.com> I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. who can help me? thanks. Best regards Daniel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/90dc1dee/attachment.html From aviv at sent.com Thu Nov 6 12:09:29 2014 From: aviv at sent.com (Aviv Shaham) Date: Thu, 06 Nov 2014 02:09:29 -0700 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415306341755.98271@nexxuspg.com> References: <1415306341755.98271@nexxuspg.com> Message-ID: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> Hi Manish, Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo. Aviv On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: > Hi, > > I have make a SIP Trunking (gateway) in FreeSwitch for connecting > Nexmo via bridge. I have added this Nexmo file under > "*\FreeSWITCH\conf\sip_profiles\external*" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: > > > Name Type Data State > ================================================================================================ > external-ipv6 profile > sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) > 192.168.1.140 alias internal ALIASED external profile > sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com > gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway > sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile > sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) > internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) > ================================================================================================ > 4 profiles 1 alias > > But when I send the request to FreeSwitch by Dial command as: > * type="xml/freeswitch-httapi"> applicati**on="set" data="sip_h_api_key=b9c280dd" /> application="set" data="sip_h_**api_secret=7678b8c4" /> application="set" data="sip_h_to=919818753995" /**> *caller**-id-number="18188535351" context="default" Dialplan="XML" > >919818753995* > > > here, *18188535351* = Nexmo virtual number for connecting call. > *919818753995* = mobile number where I am looking for making a call. > > It will not connected to Nexmo and call will be terminated with > message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 > Originate Failed. Cause: NORMAL_UNSPECIFIED > > Please find the FreeSwitch call Log and Nexmo Gateway (which I have > added in freeswitch conf external folder) as an attachment. > > Please let me know whether I am doing SIP trunking in correct way or > need to change something. > > Also, Please suggest me what will be my next step for making a call on > mobile by this ways. > > Thanks, > > Regards, Manish Talwar > > ___________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites http://www.freeswitch.org > http://confluence.freeswitch.org http://www.cluecon.com > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Email had 2 attachments: > * FsCall.txt 15k (text/plain) > * Nexmo.xml 3k (text/xml) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/b1dc072f/attachment.html From aqsyounas at gmail.com Thu Nov 6 15:31:43 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 6 Nov 2014 17:31:43 +0500 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: Sorry, Yup AAC format. On 6 November 2014 02:31, Brian West wrote: > I think you mean AAC? > > On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: > >> Hi, >> I am new to freeswitch. Does anybody know freeswitch support acc format? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/ae6bd630/attachment.html From jcabezas at inovax.com.br Thu Nov 6 15:50:17 2014 From: jcabezas at inovax.com.br (Julio Cabezas) Date: Thu, 06 Nov 2014 10:50:17 -0200 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs Message-ID: <545B6E89.2080009@inovax.com.br> Hi, How to make FreeSWITCH Sofia UA request several SIP registrations (with different AORs) to an already deployed main SIP server ? I need that for that server forward some calls to FS. Thanks. From joelewhite at gmail.com Thu Nov 6 16:33:01 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 6 Nov 2014 08:33:01 -0500 Subject: [Freeswitch-users] PostgreSQL - p_key In-Reply-To: References: <543F3C95.5050807@quentustech.com> Message-ID: Cool, thanks for the feedback :) On Wed, Nov 5, 2014 at 5:59 PM, Chris Tunbridge wrote: > if you had 100 years of calls, having say 4 channels per call, you could > have 63,173,781,074,347 calls per day > > that's 63 trillion calls... i wouldn't worry too much about seeing > repeat/cycle anytime soon. > > You are talking about having 9 quintillion records, it should be safe to > say that with bigint/serial setup like that you will be fine... you're much > more likely to have a database server crash than run out of indexes at that > point. > > On Tue, Nov 4, 2014 at 11:03 AM, Joel White wrote: > >> Well here is what I have done, I think that it should work. The PK is a >> SERIAL DataType and I went into the sequence properties and set it to >> CYCLED. The max value is 9223372036854775807 and will re-cycle once that >> number is reached. >> >> I only added this to the channels table right now. I have not seen any >> issues yet. >> >> On Mon, Nov 3, 2014 at 4:50 PM, Michael Jerris wrote: >> >>> I've used bigint with a default of a sequence in the past to good >>> results. make sure to add the option to not recreate the tables so we >>> don't mess up your fields when we change scheme and when upgrading review >>> for schema changes and make them manually >>> >>> >>> On Monday, November 3, 2014, Joel White wrote: >>> >>>> Is there another datatype that would be more suitable. I plan to have >>>> massive amounts of traffic. Would the time datatype be more suitable? >>>> >>>> >>>> >>>> On Mon, Oct 20, 2014 at 12:26 PM, Joel White >>>> wrote: >>>> >>>>> Thank you Chris >>>>> >>>>> On Fri, Oct 17, 2014 at 2:08 PM, Chris Tunbridge >>>>> wrote: >>>>> >>>>>> Serial is identical to mysql's Auto Increment, so i'm going to say >>>>>> yes. >>>>>> >>>>>> On Fri, Oct 17, 2014 at 9:36 AM, Joel White >>>>>> wrote: >>>>>> >>>>>>> Chris and William, >>>>>>> >>>>>>> Can those added PKs have a data type of SERIAL? >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Oct 15, 2014 at 11:33 PM, William King < >>>>>>> william.king at quentustech.com> wrote: >>>>>>> >>>>>>>> Yes, you can add columns to those tables with no issues. Try it >>>>>>>> and see. >>>>>>>> >>>>>>>> William King >>>>>>>> Senior Engineer >>>>>>>> Quentus Technologies, INC >>>>>>>> 1037 NE 65th St Suite 273 >>>>>>>> Seattle, WA 98115 >>>>>>>> Main: (877) 211-9337 >>>>>>>> Office: (206) 388-4772 >>>>>>>> Cell: (253) 686-5518william.king at quentustech.com >>>>>>>> >>>>>>>> On 10/8/14 9:44 AM, Joel White wrote: >>>>>>>> >>>>>>>> Is it feasible to add Primary Keys to all FreeSWITCH tables or >>>>>>>> would this break some functionality? >>>>>>>> >>>>>>>> I am asking because I am working on a multi-master replication >>>>>>>> setup >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://confluence.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/4e697670/attachment.html From mike at jerris.com Thu Nov 6 16:49:47 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 08:49:47 -0500 Subject: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge In-Reply-To: <02fb01cff990$fb89c270$f29d4750$@163.com> References: <02fb01cff990$fb89c270$f29d4750$@163.com> Message-ID: What specifically are you looking to control? > On Nov 6, 2014, at 2:12 AM, Daniel Kou wrote: > > I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. > who can help me? thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/9a5abb89/attachment.html From mike at jerris.com Thu Nov 6 16:51:38 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 08:51:38 -0500 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: <545B6E89.2080009@inovax.com.br> References: <545B6E89.2080009@inovax.com.br> Message-ID: <41734430-C624-455A-8963-44F6A551421B@jerris.com> The mod_sofia concept that makes outbound registrations is called a gateway. > On Nov 6, 2014, at 7:50 AM, Julio Cabezas wrote: > > Hi, > How to make FreeSWITCH Sofia UA request several SIP registrations > (with different AORs) to an already deployed main SIP server ? > I need that for that server forward some calls to FS. > Thanks. From steveayre at gmail.com Thu Nov 6 16:56:47 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 6 Nov 2014 13:56:47 +0000 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: <545B6E89.2080009@inovax.com.br> References: <545B6E89.2080009@inovax.com.br> Message-ID: https://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Gateway On 6 November 2014 12:50, Julio Cabezas wrote: > Hi, > How to make FreeSWITCH Sofia UA request several SIP registrations > (with different AORs) to an already deployed main SIP server ? > I need that for that server forward some calls to FS. > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/2c804319/attachment.html From joelewhite at gmail.com Thu Nov 6 17:07:32 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 6 Nov 2014 09:07:32 -0500 Subject: [Freeswitch-users] Updating via GIT Message-ID: I am having an issue with updating via GIT. I am currently on 1.4.9, but would like to upgrade to the latest 1.4.13 . I have done a GIT pull and did not see the merge error that Ken had mentioned. When I run make current, it goes through the motions and looks to be updating the installation. When I connect back to FS and issue the version command, it still states 1.4.9 How do I correct this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/a88e6a72/attachment.html From brian at freeswitch.org Thu Nov 6 17:11:25 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 08:11:25 -0600 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: It currently doesn't, I would honestly convert everything to wav files and lower your CPU requirements over AAC, Most AAC libs are not license compatible... Last we checked the API was just a mess to use. On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas wrote: > Sorry, Yup AAC format. > > On 6 November 2014 02:31, Brian West wrote: > >> I think you mean AAC? >> >> On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: >> >>> Hi, >>> I am new to freeswitch. Does anybody know freeswitch support acc format? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/6fed2103/attachment.html From ifink at machshevet.com Thu Nov 6 16:52:22 2014 From: ifink at machshevet.com (=?utf-8?Q?Israel_Fink?=) Date: Thu, 6 Nov 2014 13:52:22 +0000 Subject: [Freeswitch-users] =?utf-8?q?Confusing_about_FreeSWITCH_callID_va?= =?utf-8?q?riables?= Message-ID: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> I'm a bit confusing about the channel variables that represent the call ID. I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID 3) variable_sip_call_id, what is the different between this. The value of Channel-Call-UUID seems to be stable during the entire call, the value of Unique-ID in the begin it is the same as Channel-Call-UUID but when bridging it changes it value to another ID, then it comes back to the beginning value, i.m not clear when and why. The variable variable_sip_call_id many time don't have any value, and also it changes the value when bridging and then comes back to the previous value. I have looked for an explanation in FreeSWITCH wiki, but don't find. In the mailing list a have found an explanation here: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, but still not enough. Can someone give a good explanation about this variables,what is their purpose, and so on. Israel Fink - Developer Machshevet team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/61161fa9/attachment-0001.html From brian at freeswitch.org Thu Nov 6 17:32:04 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 08:32:04 -0600 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: References: Message-ID: What does it say when you do a git pull? On Thu, Nov 6, 2014 at 8:07 AM, Joel White wrote: > I am having an issue with updating via GIT. I am currently on 1.4.9, but > would like to upgrade to the latest 1.4.13 . > > I have done a GIT pull and did not see the merge error that Ken had > mentioned. When I run make current, it goes through the motions and looks > to be updating the installation. When I connect back to FS and issue the > version command, it still states 1.4.9 > > How do I correct this? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/6f86c660/attachment.html From krice at freeswitch.org Thu Nov 6 17:59:54 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 06 Nov 2014 08:59:54 -0600 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: Message-ID: Got ahead and update your remote and make sure you clear things out git remote set-url origin https://freeswitch.org/stash/scm/fs/freeswitch.git Then make sure you are on the head of the branch git checkout v1.4 git pull Then make current, if that doesn?t work do these commands but you?ll have to start over like a fresh checkout git clean ?fdx git reset ?-hard origin/v1.4 git pull Then ./boostrap.sh ?j && ./configure ?C etc as normal K On 11/6/14 8:07 AM, "Joel White" wrote: > I am having an issue with updating via GIT.? I am currently on 1.4.9, but > would like to upgrade to the latest 1.4.13 . > > I have done a GIT pull and did not see the merge error that Ken had > mentioned.? When I run make current, it goes through the motions and looks to > be updating the installation.? When I connect back to FS and issue the version > command, it still states 1.4.9 > > How do I correct this? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/0a1cc687/attachment.html From tfred31 at yahoo.com Thu Nov 6 17:50:35 2014 From: tfred31 at yahoo.com (T Fred Farmington) Date: Thu, 6 Nov 2014 06:50:35 -0800 Subject: [Freeswitch-users] Newbie question: Set up to answer new phone number (please be gentle) Message-ID: <1415285435.32669.YahooMailBasic@web160204.mail.bf1.yahoo.com> Since I am a total newbie to FreeSwitch please be patient and 'gentle' with me.... I have 'inherited' 2 FreeSwitch installations. One is working (version 1.0.7) and one (version 1.5.14) is a new test environment that is not working. I cannot get the new test environment to answer the inbound call and in the FreeSwitch log I find: Rejected by acl "domains" I have tried to search the FreeSwitch documentation for an answer and got lost in it. And I have tried to search the web for an answer, but did not find anything clearly spelled out. And, lastly, I tried to investigate the working installation, but could not find my answer. I have a dedicated SIP phone line which can call into the new test environment. And via FS_CLI I can see the call being received. However the connection is never established and in the log I find: Rejected by acl "domains" In both FreeSwitch environments I have a directory named: \Debug into which the previous individual installed all of the configuration parameters. Questions: 1. Where/How is FreeSwitch configured to recognize this directory and look for its parameters there? **** Maybe the non-working environment does not have that setting established and therefore is ignoring settings made there. 2. I have configured the new inbound IP address into the file: acl.conf.xml and have 'mirrored' the working environment in that regard. **** But the 'Rejected' message in the log suggests that somehow that is not being recognized. 3. Are there other files which I have missed in trying to set this up? Thanks From steveayre at gmail.com Thu Nov 6 18:11:00 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 6 Nov 2014 15:11:00 +0000 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: References: Message-ID: Try a new git clone from the new url On 6 November 2014 14:07, Joel White wrote: > I am having an issue with updating via GIT. I am currently on 1.4.9, but > would like to upgrade to the latest 1.4.13 . > > I have done a GIT pull and did not see the merge error that Ken had > mentioned. When I run make current, it goes through the motions and looks > to be updating the installation. When I connect back to FS and issue the > version command, it still states 1.4.9 > > How do I correct this? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/a2accc58/attachment.html From doron at lexifone.com Thu Nov 6 18:19:10 2014 From: doron at lexifone.com (Doron Kruh) Date: Thu, 6 Nov 2014 17:19:10 +0200 Subject: [Freeswitch-users] By pass only video rtp Message-ID: Hello, Is there a way to bypass media only of the Video RTP stream and still process the audio RTP of a video call through Freeswitch? Thanks, Doron. -- Doron Kruh, R&D Team Leader Lexifone , Hapalyam 7 Haifa. Mobile: 972-52-3823361 Office: 972-4-9126793 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/19b836ae/attachment.html From sertys at gmail.com Thu Nov 6 18:28:11 2014 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 6 Nov 2014 16:28:11 +0100 Subject: [Freeswitch-users] GIT down? Message-ID: I am getting : git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git Initialized empty Git repository in /usr/local/src/freeswitch/.git/ git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out fatal: unable to connect a socket (Connection timed out) on 2 separate machines on different networks. Is it possible the git repo is down? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/76328b3c/attachment-0001.html From rossbcan at gmail.com Thu Nov 6 18:31:03 2014 From: rossbcan at gmail.com (Bill Ross) Date: Thu, 6 Nov 2014 10:31:03 -0500 Subject: [Freeswitch-users] Recommended FS SOHO configuration, test scripts Message-ID: <050701cff9d6$ad0c5d60$07251820$@gmail.com> Folks; I am creating a hybrid OpenWrt (recent trunk) / Freeswitch-1.4.12 office in a box distribution intended for SOHO's, small business and, end users looking for secure telephony (ZRTP) solution. I am fine tuning the configuration and have a Freeswitch guru question, requiring knowledge of how freeswitch is used in the field. In particular, which modules I should compile freeswitch with. My current (compile successfully, not fully tested) modules.conf is here: http://www.rossco.org/Files/modules.conf-1.4.production.txt any suggestions regarding module additions / deletions would be greatly appreciated. Also, any pointers to scripts for automated feature / load / regression testing would be greatly appreciated. Thanks; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/668d3d74/attachment.html From aqsyounas at gmail.com Thu Nov 6 19:08:53 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 6 Nov 2014 21:08:53 +0500 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: Thank You very much.! On 6 November 2014 19:11, Brian West wrote: > It currently doesn't, I would honestly convert everything to wav files and > lower your CPU requirements over AAC, Most AAC libs are not license > compatible... Last we checked the API was just a mess to use. > > On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas wrote: > >> Sorry, Yup AAC format. >> >> On 6 November 2014 02:31, Brian West wrote: >> >>> I think you mean AAC? >>> >>> On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: >>> >>>> Hi, >>>> I am new to freeswitch. Does anybody know freeswitch support acc >>>> format? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/4fdeb12a/attachment.html From joelewhite at gmail.com Thu Nov 6 19:57:23 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 6 Nov 2014 11:57:23 -0500 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: References: Message-ID: git pull shows this remote: Counting objects: 2381, done. remote: Compressing objects: 100% (929/929), done. remote: Total 1445 (delta 1211), reused 566 (delta 504) Receiving objects: 100% (1445/1445), 238.98 KiB, done. Resolving deltas: 100% (1211/1211), completed with 368 local objects. >From https://stash.freeswitch.org/scm/fs/freeswitch ae069dc..b942d0f v1.4 -> origin/v1.4 dba9abc..7cc5209 fs-video -> origin/fs-video * [new branch] fs-video2 -> origin/fs-video2 8258180..b0050f5 master -> origin/master * [new branch] mod_smpp34 -> origin/mod_smpp34 7512761..6a69eae v1.2.stable -> origin/v1.2.stable >From https://stash.freeswitch.org/scm/fs/freeswitch * [new tag] v1.4.10 -> v1.4.10 * [new tag] v1.4.11 -> v1.4.11 * [new tag] v1.4.12 -> v1.4.12 * [new tag] v1.5.14 -> v1.5.14 Updating ae069dc..b942d0f Fast-forward Freeswitch.2012.sln | 61 + Makefile.am | 14 +- build/Makefile.centos5 | 4 +- build/Makefile.centos6 | 4 +- build/Makefile.openbsd | 6 +- build/Makefile.solaris11 | 4 +- conf/vanilla/autoload_configs/timezones.conf.xml | 1519 ++++++++++++++++++-- conf/vanilla/autoload_configs/xml_rpc.conf.xml | 6 + conf/vanilla/lang/de/vm/sounds.xml | 6 +- conf/vanilla/lang/en/vm/sounds.xml | 6 +- conf/vanilla/lang/he/vm/sounds.xml | 6 +- conf/vanilla/lang/ru/vm/sounds.xml | 6 +- conf/vanilla/vars.xml | 16 + configure.ac | 77 +- debian/bootstrap.sh | 22 +- debian/control-modules | 1 + debian/util.sh | 64 +- html5/verto/demo/img/cc_banner.gif | Bin 0 -> 36820 bytes html5/verto/demo/img/cc_banner2014.gif | Bin 42775 -> 0 bytes html5/verto/demo/img/cc_banner2014.jpg | Bin 791929 -> 0 bytes html5/verto/demo/index.html | 4 +- html5/verto/demo/js/verto-min.js | Bin 52404 -> 52560 bytes html5/verto/demo/verto.js | 7 +- html5/verto/js/src/jquery.FSRTC.js | 11 +- html5/verto/js/src/jquery.verto.js | 2 + libs/.gitignore | 1 + libs/esl/Makefile.am | 4 + libs/esl/fs_cli.c | 5 +- libs/esl/managed/ManagedEsl.2012.csproj | 2 +- libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.c | 2 + libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c | 2 +- libs/libdingaling/src/libdingaling.c | 13 +- libs/libdingaling/src/libdingaling.h | 1 + libs/sofia-sip/.update | 2 +- libs/sofia-sip/libsofia-sip-ua/sdp/sdp_parse.c | 8 +- libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su.h | 2 +- .../libsofia-sip-ua/su/sofia-sip/su_wait.h | 2 +- libs/sofia-sip/libsofia-sip-ua/su/su.c | 17 +- libs/sofia-sip/libsofia-sip-ua/su/su_wait.c | 33 +- libs/sofia-sip/libsofia-sip-ua/tport/ws.c | 12 +- libs/srtp/libsrtp.2010.vcxproj | 5 + libs/srtp/libsrtp.2012.vcxproj | 5 + libs/win32/openssl/libeay32.2010.vcxproj | 28 +- libs/win32/openssl/ssleay32.2010.vcxproj | 4 +- scripts/perl/timezone-gen.pl | 9 +- src/fs_encode.c | 23 +- src/include/switch_channel.h | 3 +- src/include/switch_core_media.h | 5 +- src/include/switch_stfu.h | 3 +- src/include/switch_types.h | 17 +- src/include/switch_utils.h | 1 + src/mod/applications/mod_commands/mod_commands.c | 25 +- .../applications/mod_conference/mod_conference.c | 35 +- src/mod/applications/mod_directory/mod_directory.c | 2 +- src/mod/applications/mod_snom/mod_snom.c | 38 +- .../mod_valet_parking/mod_valet_parking.c | 2 +- src/mod/codecs/mod_g729/Makefile.am | 4 +- src/mod/codecs/mod_opus/Makefile.am | 4 +- src/mod/codecs/mod_opus/mod_opus.c | 28 +- src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj | 135 ++ src/mod/directories/mod_ldap/Makefile.am | 2 +- src/mod/endpoints/mod_dingaling/mod_dingaling.c | 9 +- src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp | 34 + src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj | 135 ++ src/mod/endpoints/mod_sofia/mod_sofia.c | 87 +- src/mod/endpoints/mod_sofia/mod_sofia.h | 22 + src/mod/endpoints/mod_sofia/sofia.c | 318 +++- src/mod/endpoints/mod_sofia/sofia_glue.c | 26 +- src/mod/endpoints/mod_sofia/sofia_media.c | 8 +- src/mod/endpoints/mod_sofia/sofia_presence.c | 22 +- src/mod/endpoints/mod_sofia/sofia_reg.c | 97 +- src/mod/endpoints/mod_verto/mcast/mcast.c | 8 + src/mod/endpoints/mod_verto/mcast/mcast.h | 20 + src/mod/endpoints/mod_verto/mcast/mcast_cpp.cpp | 2 + src/mod/endpoints/mod_verto/mod_verto.2012.vcxproj | 27 +- src/mod/endpoints/mod_verto/mod_verto.c | 67 +- src/mod/endpoints/mod_verto/mod_verto.h | 18 +- src/mod/endpoints/mod_verto/ws.c | 16 +- src/mod/endpoints/mod_verto/ws.h | 10 +- src/mod/event_handlers/mod_rayo/mod_rayo.c | 75 +- .../mod_rayo/rayo_record_component.c | 8 +- src/mod/event_handlers/mod_rayo/srgs.c | 6 +- src/mod/event_handlers/mod_rayo/test_srgs/main.c | 9 +- src/mod/formats/mod_shout/mod_shout.c | 7 +- src/mod/languages/mod_python/mod_python.c | 2 + src/mod/xml_int/mod_xml_ldap/Makefile.am | 2 +- src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c | 16 + src/switch_channel.c | 66 +- src/switch_core_io.c | 12 + src/switch_core_media.c | 746 +++++++--- src/switch_core_session.c | 2 +- src/switch_ivr.c | 26 +- src/switch_ivr_async.c | 105 +- src/switch_ivr_play_say.c | 27 + src/switch_pgsql.c | 2 +- src/switch_rtp.c | 32 +- src/switch_stfu.c | 246 +++- src/switch_utils.c | 19 + support-d/.bashrc | 4 +- 99 files changed, 3891 insertions(+), 777 deletions(-) create mode 100644 html5/verto/demo/img/cc_banner.gif delete mode 100644 html5/verto/demo/img/cc_banner2014.gif delete mode 100644 html5/verto/demo/img/cc_banner2014.jpg create mode 100644 src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj create mode 100644 src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj On Thu, Nov 6, 2014 at 10:11 AM, Steven Ayre wrote: > Try a new git clone from the new url > > On 6 November 2014 14:07, Joel White wrote: > >> I am having an issue with updating via GIT. I am currently on 1.4.9, but >> would like to upgrade to the latest 1.4.13 . >> >> I have done a GIT pull and did not see the merge error that Ken had >> mentioned. When I run make current, it goes through the motions and looks >> to be updating the installation. When I connect back to FS and issue the >> version command, it still states 1.4.9 >> >> How do I correct this? >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/2704f61f/attachment-0001.html From joelewhite at gmail.com Thu Nov 6 19:58:15 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 6 Nov 2014 11:58:15 -0500 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: References: Message-ID: I will update the url Ken. What I had was url = https://stash.freeswitch.org/scm/fs/freeswitch.git Thank you On Thu, Nov 6, 2014 at 11:57 AM, Joel White wrote: > git pull shows this > > remote: Counting objects: 2381, done. > remote: Compressing objects: 100% (929/929), done. > remote: Total 1445 (delta 1211), reused 566 (delta 504) > Receiving objects: 100% (1445/1445), 238.98 KiB, done. > Resolving deltas: 100% (1211/1211), completed with 368 local objects. > From https://stash.freeswitch.org/scm/fs/freeswitch > ae069dc..b942d0f v1.4 -> origin/v1.4 > dba9abc..7cc5209 fs-video -> origin/fs-video > * [new branch] fs-video2 -> origin/fs-video2 > 8258180..b0050f5 master -> origin/master > * [new branch] mod_smpp34 -> origin/mod_smpp34 > 7512761..6a69eae v1.2.stable -> origin/v1.2.stable > From https://stash.freeswitch.org/scm/fs/freeswitch > * [new tag] v1.4.10 -> v1.4.10 > * [new tag] v1.4.11 -> v1.4.11 > * [new tag] v1.4.12 -> v1.4.12 > * [new tag] v1.5.14 -> v1.5.14 > Updating ae069dc..b942d0f > Fast-forward > Freeswitch.2012.sln | 61 + > Makefile.am | 14 +- > build/Makefile.centos5 | 4 +- > build/Makefile.centos6 | 4 +- > build/Makefile.openbsd | 6 +- > build/Makefile.solaris11 | 4 +- > conf/vanilla/autoload_configs/timezones.conf.xml | 1519 > ++++++++++++++++++-- > conf/vanilla/autoload_configs/xml_rpc.conf.xml | 6 + > conf/vanilla/lang/de/vm/sounds.xml | 6 +- > conf/vanilla/lang/en/vm/sounds.xml | 6 +- > conf/vanilla/lang/he/vm/sounds.xml | 6 +- > conf/vanilla/lang/ru/vm/sounds.xml | 6 +- > conf/vanilla/vars.xml | 16 + > configure.ac | 77 +- > debian/bootstrap.sh | 22 +- > debian/control-modules | 1 + > debian/util.sh | 64 +- > html5/verto/demo/img/cc_banner.gif | Bin 0 -> 36820 bytes > html5/verto/demo/img/cc_banner2014.gif | Bin 42775 -> 0 bytes > html5/verto/demo/img/cc_banner2014.jpg | Bin 791929 -> 0 > bytes > html5/verto/demo/index.html | 4 +- > html5/verto/demo/js/verto-min.js | Bin 52404 -> 52560 > bytes > html5/verto/demo/verto.js | 7 +- > html5/verto/js/src/jquery.FSRTC.js | 11 +- > html5/verto/js/src/jquery.verto.js | 2 + > libs/.gitignore | 1 + > libs/esl/Makefile.am | 4 + > libs/esl/fs_cli.c | 5 +- > libs/esl/managed/ManagedEsl.2012.csproj | 2 +- > libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.c | 2 + > libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c | 2 +- > libs/libdingaling/src/libdingaling.c | 13 +- > libs/libdingaling/src/libdingaling.h | 1 + > libs/sofia-sip/.update | 2 +- > libs/sofia-sip/libsofia-sip-ua/sdp/sdp_parse.c | 8 +- > libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su.h | 2 +- > .../libsofia-sip-ua/su/sofia-sip/su_wait.h | 2 +- > libs/sofia-sip/libsofia-sip-ua/su/su.c | 17 +- > libs/sofia-sip/libsofia-sip-ua/su/su_wait.c | 33 +- > libs/sofia-sip/libsofia-sip-ua/tport/ws.c | 12 +- > libs/srtp/libsrtp.2010.vcxproj | 5 + > libs/srtp/libsrtp.2012.vcxproj | 5 + > libs/win32/openssl/libeay32.2010.vcxproj | 28 +- > libs/win32/openssl/ssleay32.2010.vcxproj | 4 +- > scripts/perl/timezone-gen.pl | 9 +- > src/fs_encode.c | 23 +- > src/include/switch_channel.h | 3 +- > src/include/switch_core_media.h | 5 +- > src/include/switch_stfu.h | 3 +- > src/include/switch_types.h | 17 +- > src/include/switch_utils.h | 1 + > src/mod/applications/mod_commands/mod_commands.c | 25 +- > .../applications/mod_conference/mod_conference.c | 35 +- > src/mod/applications/mod_directory/mod_directory.c | 2 +- > src/mod/applications/mod_snom/mod_snom.c | 38 +- > .../mod_valet_parking/mod_valet_parking.c | 2 +- > src/mod/codecs/mod_g729/Makefile.am | 4 +- > src/mod/codecs/mod_opus/Makefile.am | 4 +- > src/mod/codecs/mod_opus/mod_opus.c | 28 +- > src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj | 135 ++ > src/mod/directories/mod_ldap/Makefile.am | 2 +- > src/mod/endpoints/mod_dingaling/mod_dingaling.c | 9 +- > src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp | 34 + > src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj | 135 ++ > src/mod/endpoints/mod_sofia/mod_sofia.c | 87 +- > src/mod/endpoints/mod_sofia/mod_sofia.h | 22 + > src/mod/endpoints/mod_sofia/sofia.c | 318 +++- > src/mod/endpoints/mod_sofia/sofia_glue.c | 26 +- > src/mod/endpoints/mod_sofia/sofia_media.c | 8 +- > src/mod/endpoints/mod_sofia/sofia_presence.c | 22 +- > src/mod/endpoints/mod_sofia/sofia_reg.c | 97 +- > src/mod/endpoints/mod_verto/mcast/mcast.c | 8 + > src/mod/endpoints/mod_verto/mcast/mcast.h | 20 + > src/mod/endpoints/mod_verto/mcast/mcast_cpp.cpp | 2 + > src/mod/endpoints/mod_verto/mod_verto.2012.vcxproj | 27 +- > src/mod/endpoints/mod_verto/mod_verto.c | 67 +- > src/mod/endpoints/mod_verto/mod_verto.h | 18 +- > src/mod/endpoints/mod_verto/ws.c | 16 +- > src/mod/endpoints/mod_verto/ws.h | 10 +- > src/mod/event_handlers/mod_rayo/mod_rayo.c | 75 +- > .../mod_rayo/rayo_record_component.c | 8 +- > src/mod/event_handlers/mod_rayo/srgs.c | 6 +- > src/mod/event_handlers/mod_rayo/test_srgs/main.c | 9 +- > src/mod/formats/mod_shout/mod_shout.c | 7 +- > src/mod/languages/mod_python/mod_python.c | 2 + > src/mod/xml_int/mod_xml_ldap/Makefile.am | 2 +- > src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c | 16 + > src/switch_channel.c | 66 +- > src/switch_core_io.c | 12 + > src/switch_core_media.c | 746 +++++++--- > src/switch_core_session.c | 2 +- > src/switch_ivr.c | 26 +- > src/switch_ivr_async.c | 105 +- > src/switch_ivr_play_say.c | 27 + > src/switch_pgsql.c | 2 +- > src/switch_rtp.c | 32 +- > src/switch_stfu.c | 246 +++- > src/switch_utils.c | 19 + > support-d/.bashrc | 4 +- > 99 files changed, 3891 insertions(+), 777 deletions(-) > create mode 100644 html5/verto/demo/img/cc_banner.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.jpg > create mode 100644 src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj > create mode 100644 src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj > > > > On Thu, Nov 6, 2014 at 10:11 AM, Steven Ayre wrote: > >> Try a new git clone from the new url >> >> On 6 November 2014 14:07, Joel White wrote: >> >>> I am having an issue with updating via GIT. I am currently on 1.4.9, >>> but would like to upgrade to the latest 1.4.13 . >>> >>> I have done a GIT pull and did not see the merge error that Ken had >>> mentioned. When I run make current, it goes through the motions and looks >>> to be updating the installation. When I connect back to FS and issue the >>> version command, it still states 1.4.9 >>> >>> How do I correct this? >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/2ae56e79/attachment-0001.html From Peter.Stevens at bbc.co.uk Thu Nov 6 20:05:57 2014 From: Peter.Stevens at bbc.co.uk (Peter Stevens) Date: Thu, 6 Nov 2014 17:05:57 +0000 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: , Message-ID: <2F938AAE444ADB47AFEE6307955A30CE2EB7BBBD@BGB01XUD1002.national.core.bbc.co.uk> Are you trying to do something with the AAC stream within freeswitch itself? If not you, could use the proxy_media (https://wiki.freeswitch.org/wiki/Proxy_Media) or bypass_media (https://wiki.freeswitch.org/wiki/Bypass_Media) modes, depending upon what you are wanting to do. We've used these to support the higher quality codecs (not supported by FS), used within the broadcast environment for journalist contributions. Peter ________________________________ From: Brian West [brian at freeswitch.org] Sent: 06 November 2014 14:11 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Does freeswitch support acc format? It currently doesn't, I would honestly convert everything to wav files and lower your CPU requirements over AAC, Most AAC libs are not license compatible... Last we checked the API was just a mess to use. On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas > wrote: Sorry, Yup AAC format. On 6 November 2014 02:31, Brian West > wrote: I think you mean AAC? On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas > wrote: Hi, I am new to freeswitch. Does anybody know freeswitch support acc format? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/578d4fad/attachment.html From mike at jerris.com Thu Nov 6 20:12:02 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 12:12:02 -0500 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: the correct new git url is https://freeswitch.org/stash/scm/fs/freeswitch.git > On Nov 6, 2014, at 10:28 AM, Daniel Ivanov wrote: > > I am getting : > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > git.freeswitch.org [0: 209.105.235.6]: errno=Connection timed out > fatal: unable to connect a socket (Connection timed out) > > on 2 separate machines on different networks. > > Is it possible the git repo is down? > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/64c48bdc/attachment.html From mike at jerris.com Thu Nov 6 20:12:46 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 12:12:46 -0500 Subject: [Freeswitch-users] By pass only video rtp In-Reply-To: References: Message-ID: <8CD4897F-A235-463B-ABCC-CC02D936117E@jerris.com> There is not at this time, and doing so would cause issues with audio/video sync. > On Nov 6, 2014, at 10:19 AM, Doron Kruh wrote: > > Hello, > > Is there a way to bypass media only of the Video RTP stream and still process the audio RTP of a video call through Freeswitch? From mike at jerris.com Thu Nov 6 20:15:46 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 12:15:46 -0500 Subject: [Freeswitch-users] Recommended FS SOHO configuration, test scripts In-Reply-To: <050701cff9d6$ad0c5d60$07251820$@gmail.com> References: <050701cff9d6$ad0c5d60$07251820$@gmail.com> Message-ID: Which modules you need very much depends on what you want to do. Generally I would not install any modules you are not explicitly using. I would use 1.4.13, it fixed some important issues found in 1.4.12 > On Nov 6, 2014, at 10:31 AM, Bill Ross wrote: > > Folks; > > I am creating a hybrid OpenWrt (recent trunk) / Freeswitch-1.4.12 office in a box distribution intended for SOHO?s, small business and, end users looking for secure telephony (ZRTP) solution. I am fine tuning the configuration and have a Freeswitch guru question, requiring knowledge of how freeswitch is used in the field. In particular, which modules I should compile freeswitch with. > > My current (compile successfully, not fully tested) modules.conf is here: > > http://www.rossco.org/Files/modules.conf-1.4.production.txt > > any suggestions regarding module additions / deletions would be greatly appreciated. > > Also, any pointers to scripts for automated feature / load / regression testing would be greatly appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/3b80f707/attachment-0001.html From brian at freeswitch.org Thu Nov 6 20:18:45 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 11:18:45 -0600 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: Seems you missed the announcement that was sent out. git remote set-url origin https://freeswitch.org/stash/scm/fs/freeswitch.git You'll be set. On Thu, Nov 6, 2014 at 9:28 AM, Daniel Ivanov wrote: > I am getting : > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out > fatal: unable to connect a socket (Connection timed out) > > on 2 separate machines on different networks. > > Is it possible the git repo is down? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/e7f97a48/attachment.html From mbodbg at gmx.net Thu Nov 6 20:20:30 2014 From: mbodbg at gmx.net (mbo) Date: Thu, 6 Nov 2014 18:20:30 +0100 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: <43E48637-62F9-467F-8AC6-47FEA9BBA879@gmx.net> The git repository has been replaced by Stash. See here: https://freeswitch.org/confluence/display/FREESWITCH/Installation Regards, Markus Am 06.11.2014 um 16:28 schrieb Daniel Ivanov : > I am getting : > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out > fatal: unable to connect a socket (Connection timed out) > > on 2 separate machines on different networks. > > Is it possible the git repo is down? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/c354f8dc/attachment.html From kris at kriskinc.com Thu Nov 6 20:31:48 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 6 Nov 2014 12:31:48 -0500 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: AAC (especially the ELD variant) is also a realtime codec used by FaceTime and others. On Thursday, November 6, 2014, Brian West wrote: > It currently doesn't, I would honestly convert everything to wav files and > lower your CPU requirements over AAC, Most AAC libs are not license > compatible... Last we checked the API was just a mess to use. > > On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas > wrote: > >> Sorry, Yup AAC format. >> >> On 6 November 2014 02:31, Brian West > > wrote: >> >>> I think you mean AAC? >>> >>> On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas >> > wrote: >>> >>>> Hi, >>>> I am new to freeswitch. Does anybody know freeswitch support acc >>>> format? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/d3a5dc6e/attachment.html From steveayre at gmail.com Thu Nov 6 21:01:01 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 6 Nov 2014 18:01:01 +0000 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: The URL has changed. https://freeswitch.org/stash/scm/fs/freeswitch.git On 6 November 2014 15:28, Daniel Ivanov wrote: > I am getting : > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out > fatal: unable to connect a socket (Connection timed out) > > on 2 separate machines on different networks. > > Is it possible the git repo is down? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/b3a85c2c/attachment-0001.html From mike at jerris.com Thu Nov 6 21:04:39 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 6 Nov 2014 13:04:39 -0500 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: References: Message-ID: <743209AA-89D4-426C-BDE6-E09AE0A156A9@jerris.com> This all looks good. Are you configuring it to a different location and somehow running a different version than this is building? > On Nov 6, 2014, at 11:57 AM, Joel White wrote: > > git pull shows this > > remote: Counting objects: 2381, done. > remote: Compressing objects: 100% (929/929), done. > remote: Total 1445 (delta 1211), reused 566 (delta 504) > Receiving objects: 100% (1445/1445), 238.98 KiB, done. > Resolving deltas: 100% (1211/1211), completed with 368 local objects. > From https://stash.freeswitch.org/scm/fs/freeswitch > ae069dc..b942d0f v1.4 -> origin/v1.4 > dba9abc..7cc5209 fs-video -> origin/fs-video > * [new branch] fs-video2 -> origin/fs-video2 > 8258180..b0050f5 master -> origin/master > * [new branch] mod_smpp34 -> origin/mod_smpp34 > 7512761..6a69eae v1.2.stable -> origin/v1.2.stable > From https://stash.freeswitch.org/scm/fs/freeswitch > * [new tag] v1.4.10 -> v1.4.10 > * [new tag] v1.4.11 -> v1.4.11 > * [new tag] v1.4.12 -> v1.4.12 > * [new tag] v1.5.14 -> v1.5.14 > Updating ae069dc..b942d0f > Fast-forward > Freeswitch.2012.sln | 61 + > Makefile.am | 14 +- > build/Makefile.centos5 | 4 +- > build/Makefile.centos6 | 4 +- > build/Makefile.openbsd | 6 +- > build/Makefile.solaris11 | 4 +- > conf/vanilla/autoload_configs/timezones.conf.xml | 1519 ++++++++++++++++++-- > conf/vanilla/autoload_configs/xml_rpc.conf.xml | 6 + > conf/vanilla/lang/de/vm/sounds.xml | 6 +- > conf/vanilla/lang/en/vm/sounds.xml | 6 +- > conf/vanilla/lang/he/vm/sounds.xml | 6 +- > conf/vanilla/lang/ru/vm/sounds.xml | 6 +- > conf/vanilla/vars.xml | 16 + > configure.ac | 77 +- > debian/bootstrap.sh | 22 +- > debian/control-modules | 1 + > debian/util.sh | 64 +- > html5/verto/demo/img/cc_banner.gif | Bin 0 -> 36820 bytes > html5/verto/demo/img/cc_banner2014.gif | Bin 42775 -> 0 bytes > html5/verto/demo/img/cc_banner2014.jpg | Bin 791929 -> 0 bytes > html5/verto/demo/index.html | 4 +- > html5/verto/demo/js/verto-min.js | Bin 52404 -> 52560 bytes > html5/verto/demo/verto.js | 7 +- > html5/verto/js/src/jquery.FSRTC.js | 11 +- > html5/verto/js/src/jquery.verto.js | 2 + > libs/.gitignore | 1 + > libs/esl/Makefile.am | 4 + > libs/esl/fs_cli.c | 5 +- > libs/esl/managed/ManagedEsl.2012.csproj | 2 +- > libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.c | 2 + > libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c | 2 +- > libs/libdingaling/src/libdingaling.c | 13 +- > libs/libdingaling/src/libdingaling.h | 1 + > libs/sofia-sip/.update | 2 +- > libs/sofia-sip/libsofia-sip-ua/sdp/sdp_parse.c | 8 +- > libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su.h | 2 +- > .../libsofia-sip-ua/su/sofia-sip/su_wait.h | 2 +- > libs/sofia-sip/libsofia-sip-ua/su/su.c | 17 +- > libs/sofia-sip/libsofia-sip-ua/su/su_wait.c | 33 +- > libs/sofia-sip/libsofia-sip-ua/tport/ws.c | 12 +- > libs/srtp/libsrtp.2010.vcxproj | 5 + > libs/srtp/libsrtp.2012.vcxproj | 5 + > libs/win32/openssl/libeay32.2010.vcxproj | 28 +- > libs/win32/openssl/ssleay32.2010.vcxproj | 4 +- > scripts/perl/timezone-gen.pl | 9 +- > src/fs_encode.c | 23 +- > src/include/switch_channel.h | 3 +- > src/include/switch_core_media.h | 5 +- > src/include/switch_stfu.h | 3 +- > src/include/switch_types.h | 17 +- > src/include/switch_utils.h | 1 + > src/mod/applications/mod_commands/mod_commands.c | 25 +- > .../applications/mod_conference/mod_conference.c | 35 +- > src/mod/applications/mod_directory/mod_directory.c | 2 +- > src/mod/applications/mod_snom/mod_snom.c | 38 +- > .../mod_valet_parking/mod_valet_parking.c | 2 +- > src/mod/codecs/mod_g729/Makefile.am | 4 +- > src/mod/codecs/mod_opus/Makefile.am | 4 +- > src/mod/codecs/mod_opus/mod_opus.c | 28 +- > src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj | 135 ++ > src/mod/directories/mod_ldap/Makefile.am | 2 +- > src/mod/endpoints/mod_dingaling/mod_dingaling.c | 9 +- > src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp | 34 + > src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj | 135 ++ > src/mod/endpoints/mod_sofia/mod_sofia.c | 87 +- > src/mod/endpoints/mod_sofia/mod_sofia.h | 22 + > src/mod/endpoints/mod_sofia/sofia.c | 318 +++- > src/mod/endpoints/mod_sofia/sofia_glue.c | 26 +- > src/mod/endpoints/mod_sofia/sofia_media.c | 8 +- > src/mod/endpoints/mod_sofia/sofia_presence.c | 22 +- > src/mod/endpoints/mod_sofia/sofia_reg.c | 97 +- > src/mod/endpoints/mod_verto/mcast/mcast.c | 8 + > src/mod/endpoints/mod_verto/mcast/mcast.h | 20 + > src/mod/endpoints/mod_verto/mcast/mcast_cpp.cpp | 2 + > src/mod/endpoints/mod_verto/mod_verto.2012.vcxproj | 27 +- > src/mod/endpoints/mod_verto/mod_verto.c | 67 +- > src/mod/endpoints/mod_verto/mod_verto.h | 18 +- > src/mod/endpoints/mod_verto/ws.c | 16 +- > src/mod/endpoints/mod_verto/ws.h | 10 +- > src/mod/event_handlers/mod_rayo/mod_rayo.c | 75 +- > .../mod_rayo/rayo_record_component.c | 8 +- > src/mod/event_handlers/mod_rayo/srgs.c | 6 +- > src/mod/event_handlers/mod_rayo/test_srgs/main.c | 9 +- > src/mod/formats/mod_shout/mod_shout.c | 7 +- > src/mod/languages/mod_python/mod_python.c | 2 + > src/mod/xml_int/mod_xml_ldap/Makefile.am | 2 +- > src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c | 16 + > src/switch_channel.c | 66 +- > src/switch_core_io.c | 12 + > src/switch_core_media.c | 746 +++++++--- > src/switch_core_session.c | 2 +- > src/switch_ivr.c | 26 +- > src/switch_ivr_async.c | 105 +- > src/switch_ivr_play_say.c | 27 + > src/switch_pgsql.c | 2 +- > src/switch_rtp.c | 32 +- > src/switch_stfu.c | 246 +++- > src/switch_utils.c | 19 + > support-d/.bashrc | 4 +- > 99 files changed, 3891 insertions(+), 777 deletions(-) > create mode 100644 html5/verto/demo/img/cc_banner.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.jpg > create mode 100644 src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj > create mode 100644 src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj > > > > On Thu, Nov 6, 2014 at 10:11 AM, Steven Ayre > wrote: > Try a new git clone from the new url > > On 6 November 2014 14:07, Joel White > wrote: > I am having an issue with updating via GIT. I am currently on 1.4.9, but would like to upgrade to the latest 1.4.13 . > > I have done a GIT pull and did not see the merge error that Ken had mentioned. When I run make current, it goes through the motions and looks to be updating the installation. When I connect back to FS and issue the version command, it still states 1.4.9 > > How do I correct this? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/18311f01/attachment-0001.html From jcabezas at inovax.com.br Thu Nov 6 20:44:22 2014 From: jcabezas at inovax.com.br (Julio Cesar Esteves Cabezas) Date: Thu, 6 Nov 2014 15:44:22 -0200 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs Message-ID: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> I have 100+ AORS to register with the main server. So, as far as I understand, I have to define 100+ gateways to the same main SIP server in FS, each one doing a single AOR registration. Is there any support for multiple AOR registrations per gateway ? Or any other more "compact" scheme ? Thanks. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, November 06, 2014 11:52 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs The mod_sofia concept that makes outbound registrations is called a gateway. > On Nov 6, 2014, at 7:50 AM, Julio Cabezas wrote: > > Hi, > How to make FreeSWITCH Sofia UA request several SIP registrations > (with different AORs) to an already deployed main SIP server ? > I need that for that server forward some calls to FS. > Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sasha at greenseedtech.com Thu Nov 6 20:50:38 2014 From: sasha at greenseedtech.com (Sasha Pachev) Date: Thu, 6 Nov 2014 10:50:38 -0700 Subject: [Freeswitch-users] mod_dptools/mod_spandsp and tone detection Message-ID: I have discovered an issue when using mod_dptools and mod_spandsp together. Both define app "stop_tone_detect". mod_dptools defines "tone_detect" to start tone detection, mod_spandsp defines "start_tone_detect". Both modules load successfully, and the conflict is resolved by awarding whoever loads the last the entry for stop_tone_detect. This creates problems - for example, if mod_spandsp wins, and the dialplan uses tone_detect/stop_tone_detect, then mod_dptools app will be called for starting, and mod_spandsp will be called for stopping, which in effect means it is a no-op. Is it considered an error to load both modules in the same configuration? What is the recommended solution/workaround if you want to use some of the functionality of mod_dptools and some of the functionality of mod_spandsp? From krice at freeswitch.org Thu Nov 6 21:14:20 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 06 Nov 2014 12:14:20 -0600 Subject: [Freeswitch-users] GIT down? In-Reply-To: Message-ID: The GIT protocol has been deprecated for quite a while now On 11/6/14 9:28 AM, "Daniel Ivanov" wrote: > I am getting :? > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > git.freeswitch.org [0: 209.105.235.6]: > errno=Connection timed out > fatal: unable to connect a socket (Connection timed out) > > on 2 separate machines on different networks. > > Is it possible the git repo is down? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/85a1751c/attachment.html From brian at freeswitch.org Thu Nov 6 21:40:41 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 12:40:41 -0600 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: References: Message-ID: I was investigating using AAC it was file formats for playback. I didn't think of it for a voice codec at the time... I could hear that... :P Guessing it would be nice and clear like OPUS/48000 On Thu, Nov 6, 2014 at 11:31 AM, Kristian Kielhofner wrote: > AAC (especially the ELD variant) is also a realtime codec used by FaceTime > and others. > > > On Thursday, November 6, 2014, Brian West wrote: > >> It currently doesn't, I would honestly convert everything to wav files >> and lower your CPU requirements over AAC, Most AAC libs are not license >> compatible... Last we checked the API was just a mess to use. >> >> On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas wrote: >> >>> Sorry, Yup AAC format. >>> >>> On 6 November 2014 02:31, Brian West wrote: >>> >>>> I think you mean AAC? >>>> >>>> On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: >>>> >>>>> Hi, >>>>> I am new to freeswitch. Does anybody know freeswitch support acc >>>>> format? >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > > > -- > Sent from mobile device > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/ffbf4a4d/attachment-0001.html From max at nysolutions.com Thu Nov 6 21:50:46 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 6 Nov 2014 18:50:46 +0000 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: Can links on this page be fixed? https://freeswitch.org/confluence/display/FREESWITCH/Debian Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, November 06, 2014 12:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] GIT down? Seems you missed the announcement that was sent out. git remote set-url origin https://freeswitch.org/stash/scm/fs/freeswitch.git You'll be set. On Thu, Nov 6, 2014 at 9:28 AM, Daniel Ivanov > wrote: I am getting : git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git Initialized empty Git repository in /usr/local/src/freeswitch/.git/ git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out fatal: unable to connect a socket (Connection timed out) on 2 separate machines on different networks. Is it possible the git repo is down? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org [http://billing.freeswitch.org/templates/default/img/whmcslogo.png] Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/7e83fcf1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/7e83fcf1/attachment.jpg From aqsyounas at gmail.com Thu Nov 6 22:12:48 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Nov 2014 00:12:48 +0500 Subject: [Freeswitch-users] Does freeswitch support acc format? In-Reply-To: <2F938AAE444ADB47AFEE6307955A30CE2EB7BBBD@BGB01XUD1002.national.core.bbc.co.uk> References: <2F938AAE444ADB47AFEE6307955A30CE2EB7BBBD@BGB01XUD1002.national.core.bbc.co.uk> Message-ID: All I want is to play a sound file as IVR in AAC format.Is there anyway to do so? Thanks for your reply. Really Appreciated. On 6 November 2014 22:05, Peter Stevens wrote: > Are you trying to do something with the AAC stream within freeswitch > itself? > > > > If not you, could use the proxy_media ( > https://wiki.freeswitch.org/wiki/Proxy_Media) or bypass_media ( > https://wiki.freeswitch.org/wiki/Bypass_Media) modes, depending upon what > you are wanting to do. > > > > We've used these to support the higher quality codecs (not supported by > FS), used within the broadcast environment for journalist contributions. > > > > Peter > > > > > ------------------------------ > *From:* Brian West [brian at freeswitch.org] > *Sent:* 06 November 2014 14:11 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Does freeswitch support acc format? > > It currently doesn't, I would honestly convert everything to wav files > and lower your CPU requirements over AAC, Most AAC libs are not license > compatible... Last we checked the API was just a mess to use. > > On Thu, Nov 6, 2014 at 6:31 AM, Aqs Younas wrote: > >> Sorry, Yup AAC format. >> >> On 6 November 2014 02:31, Brian West wrote: >> >>> I think you mean AAC? >>> >>> On Wed, Nov 5, 2014 at 2:30 PM, Aqs Younas wrote: >>> >>>> Hi, >>>> I am new to freeswitch. Does anybody know freeswitch support acc >>>> format? >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/c6772d2a/attachment-0001.html From aqsyounas at gmail.com Thu Nov 6 22:28:55 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Nov 2014 00:28:55 +0500 Subject: [Freeswitch-users] How to originate call over another server using ESL Message-ID: Hi, I want freeswitch to send a call to asterisk's extension running over some different ip using ESL. But does not know how to do so.Can anybody help? I have successfully installed ESL for python. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/6d6b6d2e/attachment.html From cmrienzo at gmail.com Thu Nov 6 22:35:31 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 6 Nov 2014 14:35:31 -0500 Subject: [Freeswitch-users] mod_dptools/mod_spandsp and tone detection In-Reply-To: References: Message-ID: If you don't need both APPs, just load the modules in the order that provides what you need until the issue gets fixed in master. On Thu, Nov 6, 2014 at 12:50 PM, Sasha Pachev wrote: > I have discovered an issue when using mod_dptools and mod_spandsp > together. Both define app "stop_tone_detect". mod_dptools defines > "tone_detect" to start tone detection, mod_spandsp defines > "start_tone_detect". Both modules load successfully, and the conflict > is resolved by awarding whoever loads the last the entry for > stop_tone_detect. This creates problems - for example, if mod_spandsp > wins, and the dialplan uses tone_detect/stop_tone_detect, then > mod_dptools app will be called for starting, and mod_spandsp will be > called for stopping, which in effect means it is a no-op. > > Is it considered an error to load both modules in the same > configuration? What is the recommended solution/workaround if you want > to use some of the functionality of mod_dptools and some of the > functionality of mod_spandsp? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/d347c0bc/attachment.html From cmrienzo at gmail.com Thu Nov 6 22:52:51 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Thu, 6 Nov 2014 14:52:51 -0500 Subject: [Freeswitch-users] mod_dptools/mod_spandsp and tone detection In-Reply-To: References: Message-ID: mod_spandsp start_tone_detect and stop_tone_detect has been renamed to spandsp_start_tone_detect and spandsp_stop_tone_detect in master branch. On Thu, Nov 6, 2014 at 2:35 PM, Christopher Rienzo wrote: > If you don't need both APPs, just load the modules in the order that > provides what you need until the issue gets fixed in master. > > On Thu, Nov 6, 2014 at 12:50 PM, Sasha Pachev > wrote: > >> I have discovered an issue when using mod_dptools and mod_spandsp >> together. Both define app "stop_tone_detect". mod_dptools defines >> "tone_detect" to start tone detection, mod_spandsp defines >> "start_tone_detect". Both modules load successfully, and the conflict >> is resolved by awarding whoever loads the last the entry for >> stop_tone_detect. This creates problems - for example, if mod_spandsp >> wins, and the dialplan uses tone_detect/stop_tone_detect, then >> mod_dptools app will be called for starting, and mod_spandsp will be >> called for stopping, which in effect means it is a no-op. >> >> Is it considered an error to load both modules in the same >> configuration? What is the recommended solution/workaround if you want >> to use some of the functionality of mod_dptools and some of the >> functionality of mod_spandsp? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/74ff453a/attachment.html From brian at freeswitch.org Thu Nov 6 23:02:41 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 14:02:41 -0600 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> References: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> Message-ID: Nope. On Thu, Nov 6, 2014 at 11:44 AM, Julio Cesar Esteves Cabezas < jcabezas at inovax.com.br> wrote: > I have 100+ AORS to register with the main server. > So, as far as I understand, I have to define 100+ gateways to the same main > SIP server in FS, each one doing a single AOR registration. > Is there any support for multiple AOR registrations per gateway ? > Or any other more "compact" scheme ? > > Thanks. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Michael > Jerris > Sent: Thursday, November 06, 2014 11:52 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Registering FreeSWITCH with another server, > multiples AORs > > The mod_sofia concept that makes outbound registrations is called a > gateway. > > > On Nov 6, 2014, at 7:50 AM, Julio Cabezas > wrote: > > > > Hi, > > How to make FreeSWITCH Sofia UA request several SIP registrations > > (with different AORs) to an already deployed main SIP server ? > > I need that for that server forward some calls to FS. > > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/1ba279ee/attachment.html From brian at freeswitch.org Thu Nov 6 23:03:31 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 14:03:31 -0600 Subject: [Freeswitch-users] How to originate call over another server using ESL In-Reply-To: References: Message-ID: You'll wanna use the originate api call https://wiki.freeswitch.org/wiki/Mod_commands#originate On Thu, Nov 6, 2014 at 1:28 PM, Aqs Younas wrote: > Hi, > I want freeswitch to send a call to asterisk's extension running over some > different ip using ESL. > But does not know how to do so.Can anybody help? > > I have successfully installed ESL for python. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/fefd1157/attachment-0001.html From krice at freeswitch.org Thu Nov 6 23:12:52 2014 From: krice at freeswitch.org (Ken Rice) Date: Thu, 06 Nov 2014 14:12:52 -0600 Subject: [Freeswitch-users] GIT down? In-Reply-To: Message-ID: fixed On 11/6/14 12:50 PM, "Moishe Grunstein" wrote: > Can links on this page be fixed? > https://freeswitch.org/confluence/display/FREESWITCH/Debian > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > Service Request Email: support at nysolutions.com > Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West > Sent: Thursday, November 06, 2014 12:19 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] GIT down? > > > Seems you missed the announcement that was sent out. > > > > git remote set-url origin https://freeswitch.org/stash/scm/fs/freeswitch.git > > > > You'll be set. > > > > On Thu, Nov 6, 2014 at 9:28 AM, Daniel Ivanov wrote: > > I am getting : > > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git > > > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > > git.freeswitch.org [0: 209.105.235.6]: > errno=Connection timed out > > fatal: unable to connect a socket (Connection timed out) > > > > on 2 separate machines on different networks. > > > > Is it possible the git repo is down? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/db933709/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/db933709/attachment.jpe From brian at freeswitch.org Thu Nov 6 23:40:00 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 14:40:00 -0600 Subject: [Freeswitch-users] GIT down? In-Reply-To: References: Message-ID: On Thu, Nov 6, 2014 at 2:12 PM, Ken Rice wrote: > fixed > > > On 11/6/14 12:50 PM, "Moishe Grunstein" wrote: > > Can links on this page be fixed? > https://freeswitch.org/confluence/display/FREESWITCH/Debian > > Thanks, > > Moishe Grunstein > Tornado Computer Systems, Inc. > 212.400.7650 888.IPPBX.US > > *Service Request Email: support at nysolutions.com > *Polycom Certified VAR > Microsoft Small Business Specialist, Cisco SMB Select Certified > > Computer Networking * Managed Services * IP Video Surveillance * Network > Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network > Security * Site Surveys * CMS > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Brian West > *Sent:* Thursday, November 06, 2014 12:19 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] GIT down? > > > Seems you missed the announcement that was sent out. > > > > git remote set-url origin > https://freeswitch.org/stash/scm/fs/freeswitch.git > > > > You'll be set. > > > > On Thu, Nov 6, 2014 at 9:28 AM, Daniel Ivanov wrote: > > I am getting : > > git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git < > http://git.freeswitch.org/freeswitch.git> > > Initialized empty Git repository in /usr/local/src/freeswitch/.git/ > > git.freeswitch.org [0: 209.105.235.6]: > errno=Connection timed out > > fatal: unable to connect a socket (Connection timed out) > > > > on 2 separate machines on different networks. > > > > Is it possible the git repo is down? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > Ken > > > > *http://www.FreeSWITCH.org > http://www.ClueCon.com http://www.OSTAG.org > *irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/d8996b3c/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image.jpg Type: image/jpeg Size: 2424 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/d8996b3c/attachment-0001.jpg From anthony.minessale at gmail.com Fri Nov 7 00:07:24 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 6 Nov 2014 15:07:24 -0600 Subject: [Freeswitch-users] mod_dptools/mod_spandsp and tone detection In-Reply-To: References: Message-ID: I don't know what version of FS you are using but I do not see this. I have commented on your JIRA. https://freeswitch.org/jira/browse/FS-6966 On Thu, Nov 6, 2014 at 1:35 PM, Christopher Rienzo wrote: > If you don't need both APPs, just load the modules in the order that > provides what you need until the issue gets fixed in master. > > On Thu, Nov 6, 2014 at 12:50 PM, Sasha Pachev > wrote: > >> I have discovered an issue when using mod_dptools and mod_spandsp >> together. Both define app "stop_tone_detect". mod_dptools defines >> "tone_detect" to start tone detection, mod_spandsp defines >> "start_tone_detect". Both modules load successfully, and the conflict >> is resolved by awarding whoever loads the last the entry for >> stop_tone_detect. This creates problems - for example, if mod_spandsp >> wins, and the dialplan uses tone_detect/stop_tone_detect, then >> mod_dptools app will be called for starting, and mod_spandsp will be >> called for stopping, which in effect means it is a no-op. >> >> Is it considered an error to load both modules in the same >> configuration? What is the recommended solution/workaround if you want >> to use some of the functionality of mod_dptools and some of the >> functionality of mod_spandsp? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/83b5c606/attachment.html From igorolhovskiy at gmail.com Thu Nov 6 22:26:31 2014 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Thu, 06 Nov 2014 21:26:31 +0200 Subject: [Freeswitch-users] Verto pass variables dialplan Message-ID: <545BCB67.30603@gmail.com> Hi! Is there any way to get in dialplan (XML) parameters, passed from browser via js? In my case I appended extraHeaders. For ex, 2014-11-06 17:25:19.364834 [ALERT] mod_verto.c:1284 READ 178.54.1.162:52512 [{ "jsonrpc": "2.0", "method": "verto.invite", "params": { "sdp": "......", "dialogParams": { "useVideo": false, "useStereo": false, "tag": "webvideotag", "login": "XXXXX at webrtc-dev.webcall.today", "destination_number": "9001", "caller_id_name": "XXXXXX", "caller_id_number": "XXXXX", * "extraHeaders": {** ** "PageUrl": "https://my.webcall.today/"** ** },* "callID": "db94fb82-f113-62e5-9e61-e091715b3c4f", "remote_caller_id_name": "Outbound Call", "remote_caller_id_number": "9001" }, "sessid": "6a4c4a47-9de0-8af3-38e0-971ba1f63198" }, "id": 4 }] Is there any way to get this PageUrl parameter in dialplan variables? Freeswitch 1.5.14b -- Regards, Igor. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/93f4e1ca/attachment.html From brian at freeswitch.org Fri Nov 7 01:23:21 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 6 Nov 2014 16:23:21 -0600 Subject: [Freeswitch-users] Confusing about FreeSWITCH callID variables In-Reply-To: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> References: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> Message-ID: 1. Unique-ID is the session UUID, this won't change at all during the session (aka call). 2. Channel-Call-UUID is either the same as Unique-ID or the Unique-ID of the session you're bridged to by looking at the source code. 3. sip_call_id will be the sip call_id of an inbound/outbound SIP call, More than likely the outbound would match Unique-ID and on inbound it would be what ever the remote party sets/sends. The bigger question is what exactly are you trying to solve/understand? On Thu, Nov 6, 2014 at 7:52 AM, Israel Fink wrote: > I'm a bit confusing about the channel variables that represent the call > ID. > > I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID > 3) variable_sip_call_id, what is the different between this. > > The value of Channel-Call-UUID seems to be stable during the entire call, > the value of Unique-ID in the begin it is the same as Channel-Call-UUID but > when bridging it changes it value to another ID, then it comes back to the > beginning value, i.m not clear when and why. > > The variable variable_sip_call_id many time don't have any value, and also > it changes the value when bridging and then comes back to the previous > value. > > I have looked for an explanation in FreeSWITCH wiki, but don't find. > > In the mailing list a have found an explanation here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, > but still not enough. > Can someone give a good explanation about this variables,what is their > purpose, and so on. > > > Israel Fink - Developer > Machshevet team > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/67ecee0b/attachment.html From kris at kriskinc.com Fri Nov 7 01:46:56 2014 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 6 Nov 2014 17:46:56 -0500 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> References: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> Message-ID: This won't be as bad as you think it is. Write something to generate it statically or return the XML via mod_xml_curl and Sofia will handle it just fine. I had to lab up some crazy scenario once where I created hundreds of gateways on a profile and it worked just fine. On Thursday, November 6, 2014, Julio Cesar Esteves Cabezas < jcabezas at inovax.com.br> wrote: > I have 100+ AORS to register with the main server. > So, as far as I understand, I have to define 100+ gateways to the same main > SIP server in FS, each one doing a single AOR registration. > Is there any support for multiple AOR registrations per gateway ? > Or any other more "compact" scheme ? > > Thanks. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On > Behalf Of Michael > Jerris > Sent: Thursday, November 06, 2014 11:52 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Registering FreeSWITCH with another server, > multiples AORs > > The mod_sofia concept that makes outbound registrations is called a > gateway. > > > On Nov 6, 2014, at 7:50 AM, Julio Cabezas > wrote: > > > > Hi, > > How to make FreeSWITCH Sofia UA request several SIP registrations > > (with different AORs) to an already deployed main SIP server ? > > I need that for that server forward some calls to FS. > > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from mobile device -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/18df908a/attachment-0001.html From cmrienzo at gmail.com Fri Nov 7 01:51:12 2014 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Thu, 6 Nov 2014 17:51:12 -0500 Subject: [Freeswitch-users] mod_dptools/mod_spandsp and tone detection In-Reply-To: References: Message-ID: <47AD2252-6AEE-4CA2-8C4D-587F7D16E9B8@gmail.com> This is fixed. > On Nov 6, 2014, at 16:07, Anthony Minessale wrote: > > I don't know what version of FS you are using but I do not see this. I have commented on your JIRA. > https://freeswitch.org/jira/browse/FS-6966 > >> On Thu, Nov 6, 2014 at 1:35 PM, Christopher Rienzo wrote: >> If you don't need both APPs, just load the modules in the order that provides what you need until the issue gets fixed in master. >> >>> On Thu, Nov 6, 2014 at 12:50 PM, Sasha Pachev wrote: >>> I have discovered an issue when using mod_dptools and mod_spandsp >>> together. Both define app "stop_tone_detect". mod_dptools defines >>> "tone_detect" to start tone detection, mod_spandsp defines >>> "start_tone_detect". Both modules load successfully, and the conflict >>> is resolved by awarding whoever loads the last the entry for >>> stop_tone_detect. This creates problems - for example, if mod_spandsp >>> wins, and the dialplan uses tone_detect/stop_tone_detect, then >>> mod_dptools app will be called for starting, and mod_spandsp will be >>> called for stopping, which in effect means it is a no-op. >>> >>> Is it considered an error to load both modules in the same >>> configuration? What is the recommended solution/workaround if you want >>> to use some of the functionality of mod_dptools and some of the >>> functionality of mod_spandsp? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141106/31d6c9b6/attachment.html From jcabezas at inovax.com.br Fri Nov 7 02:17:19 2014 From: jcabezas at inovax.com.br (Julio Cabezas) Date: Thu, 06 Nov 2014 21:17:19 -0200 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: <545B6E89.2080009@inovax.com.br> References: <545B6E89.2080009@inovax.com.br> Message-ID: <545C017F.2020003@inovax.com.br> Sorry, this is a post just for testing my e-mail client thread capabilities. On 11/6/2014 10:50 AM, Julio Cabezas wrote: > Hi, > How to make FreeSWITCH Sofia UA request several SIP registrations > (with different AORs) to an already deployed main SIP server ? > I need that for that server forward some calls to FS. > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From wilddragon at 163.com Fri Nov 7 04:01:38 2014 From: wilddragon at 163.com (wilddragon at 163.com) Date: Fri, 7 Nov 2014 09:01:38 +0800 Subject: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge References: <02fb01cff990$fb89c270$f29d4750$@163.com>, Message-ID: <201411070901372157762@163.com> In this case, I can not see any video beside black picture, I'd like to know how to see the video coming from other parties,and how to control to see which party. wilddragon at 163.com From: Michael Jerris Date: 2014-11-06 21:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge What specifically are you looking to control? On Nov 6, 2014, at 2:12 AM, Daniel Kou wrote: I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. who can help me? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/e0e92b52/attachment.html From wilddragon at 163.com Fri Nov 7 05:07:27 2014 From: wilddragon at 163.com (wilddragon at 163.com) Date: Fri, 7 Nov 2014 10:07:27 +0800 Subject: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge References: <02fb01cff990$fb89c270$f29d4750$@163.com>, , <201411070901372157762@163.com> Message-ID: <201411071007258272904@163.com> OK, I set conference-flags and member-flags to video-bridge, I can see the video each other. now we can research how to decide to see which party in the bridge. wilddragon at 163.com From: wilddragon at 163.com Date: 2014-11-07 09:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge In this case, I can not see any video beside black picture, I'd like to know how to see the video coming from other parties,and how to control to see which party. wilddragon at 163.com From: Michael Jerris Date: 2014-11-06 21:49 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to use mod_conference conference-flags and member-flags: video-bridge What specifically are you looking to control? On Nov 6, 2014, at 2:12 AM, Daniel Kou wrote: I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. who can help me? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/78e3e218/attachment-0001.html From manish.talwar at nexxuspg.com Fri Nov 7 11:05:34 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Fri, 7 Nov 2014 08:05:34 +0000 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> References: <1415306341755.98271@nexxuspg.com>, <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> Message-ID: <1415396248535.99205@nexxuspg.com> Hi, Thanks for your suggestion, I have tried it and I am able to do a Inbound call via Nexmo now. But still I am not able to make any outbound call from my application. I have checked the FreeSwitch log by siptrace enable and found that my call was terminated with a SIP message as " IP/2.0 407 Proxy Authentication Required". Please find the siptrace log for my call as an attachment. and let me know what changes or configuration I need to make for Proxy Authentication Header. Thanks, Regards, Manish Talwar ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 06 November 2014 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo. Aviv On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: Hi, I have make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: 919818753995 here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/aea7cc1e/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: SipTrace.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/aea7cc1e/attachment-0001.txt From lists at telefaks.de Fri Nov 7 15:23:28 2014 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 07 Nov 2014 13:23:28 +0100 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address Message-ID: <545CB9C0.1030807@telefaks.de> Hell, we have the following problem: Our Freswitch is behind NAT. We are sending faxes to a SIP provider. Dependend on the destination number, the faxes are received or not. Faxes are always sent via the same SIP provider and the same dialplan, but I expect, they may be routed differently via other, subsequent SIP providers. Regarding the SDP I can see, that the c-line does contain the our external IP, but the o-Line does contain the local IP in the unicast_address field. We are routing the call via a defined gateway in the external profile, which has external_sip_ip set and external_rtp_ip set. Dialplan is: Here is the SDP ==================================== v=0. o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. s=FreeSWITCH. c=IN IP4 212.xxx.xxx.106. t=0 0. m=audio 12056 RTP/AVP 0 8. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=maxptime:240. ==================================== I suspect, that the following SIP providers may have a problem with the o-line with the local IP. So - Is there any way to control this? E.g. via Dialplan variable? ACL also seems to be fine 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding 192.168.0.0/16 (allow) [] to list nat.auto I see the same behaviour also in https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with proxy_media turned on" There is a link for a patch, which is no longer available. Did this go into the main release? Does anybody have this patch? -- With kind regards Peter From brian at freeswitch.org Fri Nov 7 16:16:18 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Nov 2014 07:16:18 -0600 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address In-Reply-To: <545CB9C0.1030807@telefaks.de> References: <545CB9C0.1030807@telefaks.de> Message-ID: what do you have local-network-acl, ext-rtp-ip and ext-sip-ip? On Fri, Nov 7, 2014 at 6:23 AM, Peter Steinbach wrote: > Hell, > > we have the following problem: > > Our Freswitch is behind NAT. We are sending faxes to a SIP provider. > Dependend on the destination number, the faxes are received or not. > Faxes are always sent via the same SIP provider and the same dialplan, > but I expect, they may be routed differently via other, subsequent SIP > providers. > > Regarding the SDP I can see, that the c-line does contain the our > external IP, but the o-Line does contain the local IP in the > unicast_address field. > We are routing the call via a defined gateway in the external profile, > which has external_sip_ip set and external_rtp_ip set. > Dialplan is: > > > data="sofia/gateway/QSC/06912345678 at sip.qsc.de"/> > > Here is the SDP > ==================================== > v=0. > o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. > s=FreeSWITCH. > c=IN IP4 212.xxx.xxx.106. > t=0 0. > m=audio 12056 RTP/AVP 0 8. > a=rtpmap:0 PCMU/8000/1. > a=rtpmap:8 PCMA/8000/1. > a=maxptime:240. > ==================================== > > I suspect, that the following SIP providers may have a problem with the > o-line with the local IP. > So - Is there any way to control this? E.g. via Dialplan variable? > > ACL also seems to be fine > 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding > 192.168.0.0/16 (allow) [] to list rfc1918.auto > 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > > I see the same behaviour also in > https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with > proxy_media turned on" > There is a link for a patch, which is no longer available. Did this go > into the main release? Does anybody have this patch? > > > -- > With kind regards > Peter > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/6e980799/attachment.html From brian at freeswitch.org Fri Nov 7 16:27:12 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Nov 2014 07:27:12 -0600 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: References: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> Message-ID: https://freeswitch.org/fisheye/changelog/freeswitch?cs=08ff88ec31572071c5e5dbe6fad51578814c4bd9 Not hard :) This could help. On Thu, Nov 6, 2014 at 4:46 PM, Kristian Kielhofner wrote: > This won't be as bad as you think it is. Write something to generate it > statically or return the XML via mod_xml_curl and Sofia will handle it > just fine. > > I had to lab up some crazy scenario once where I created hundreds of > gateways on a profile and it worked just fine. > > > On Thursday, November 6, 2014, Julio Cesar Esteves Cabezas < > jcabezas at inovax.com.br> wrote: > >> I have 100+ AORS to register with the main server. >> So, as far as I understand, I have to define 100+ gateways to the same >> main >> SIP server in FS, each one doing a single AOR registration. >> Is there any support for multiple AOR registrations per gateway ? >> Or any other more "compact" scheme ? >> >> Thanks. >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Michael >> Jerris >> Sent: Thursday, November 06, 2014 11:52 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Registering FreeSWITCH with another >> server, >> multiples AORs >> >> The mod_sofia concept that makes outbound registrations is called a >> gateway. >> >> > On Nov 6, 2014, at 7:50 AM, Julio Cabezas >> wrote: >> > >> > Hi, >> > How to make FreeSWITCH Sofia UA request several SIP registrations >> > (with different AORs) to an already deployed main SIP server ? >> > I need that for that server forward some calls to FS. >> > Thanks. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- > Sent from mobile device > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/f39afa90/attachment.html From matsumoto at itsherpa.com Fri Nov 7 17:02:13 2014 From: matsumoto at itsherpa.com (=?UTF-8?B?5p2+5pys56WQ5b+X?=) Date: Fri, 7 Nov 2014 23:02:13 +0900 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: Thank you Mr.Seven Du I got the sound !. it was my mistake. because of localhost. It should be IP. $con = new ESL::ESLconnection("localhost", 8021, ClueCon); However, I still have problem. Let me know how to get 'member_id' below. in perl. i tried below. $e->getHeader("Caller-Caller-ID-Name"); $e->getHeader("Caller-Caller-ID-Number"); conference play [|] thank you. On Wed, Nov 5, 2014 at 10:52 AM, Seven Du wrote: > I don?t quite understand you but you should not put SPACE between % and > s. Try to paste the real code and debug logs on pastebin to get better help. > > On Tuesday, November 4, 2014 at 11:51 PM, ???? wrote: > > Dear Stanislav Sinyagin > > Thank you for your help. > > ????????????????????????????? > > ???Perl??? > > ?????????????????????????wav?????????????????? > > ????????????????????? > my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > ???$ con-> bgapi ($ api_cmd); ???????????? > > fs_cli ? ?conference ************* play? ????????????????????????? > > ????????????? > > ??????????? > > > > On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin > wrote: > > Dear Matsumoto-san, > > I think it will be easier if you write in Japanese, then it will be clear > how we could help. I know a few Japanese-speaking colleagues who may help > in communicating. > > > > > On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > > Hello > > I have two issues. > > I am writing in Perl. > > While 2 people are talking in a conference room, the one person want to > play the sound. > > In "Caller-Username", can you get useless. > I have tried the above but, Member: it will not become a *** not found.. > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > I want to play the above in the background > It can not play in the next program. > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav); > > $ con-> bgapi ($ api_cmd); > > Best regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/f4ba641e/attachment-0001.html From aqsyounas at gmail.com Fri Nov 7 17:48:36 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Nov 2014 19:48:36 +0500 Subject: [Freeswitch-users] how to call asterisk extension from freeswitch using mod_event_socket. Message-ID: I want to call asterisk extension from freeswitch using ESL. Here is my python file. #!/usr/bin/env python import string import sys from ESL import * con = ESLconnection("localhost","8021","ClueCon") #are we connected? if con.connected: con.execute("bridge","sofia/external/555 at 192.168.1.44"); When a run this it does not generate call over asterisk extension. But when i do this using dialplan it works.Here is my default.xml *But i want to do this using mod_event_socket.* Your help would be much appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/005318ab/attachment.html From krice at freeswitch.org Fri Nov 7 18:02:41 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 07 Nov 2014 15:02:41 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <545cdf118246c_96ed10333285986@ip-10-33-128-229.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/427a93b9/attachment.html From mvar78 at gmail.com Fri Nov 7 18:15:07 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Fri, 7 Nov 2014 16:15:07 +0100 Subject: [Freeswitch-users] [MOD Nibblebill] Postpaid Customer In-Reply-To: <1415039029.95746.YahooMailNeo@web162004.mail.bf1.yahoo.com> References: <7B5A71BE-8D38-4ABF-810E-D1D154AE13F3@gmail.com> <8936CF19-235A-47E0-9E07-FAC5A4151368@freeswitch.org> <7DAF7A45-C9A8-469C-B4B3-63EC96D3CA12@gmail.com> <0000014976ba20a3-a7ec3ec2-9c92-4144-b100-6dbfa40951e9-000000@email.amazonses.com> <1415039029.95746.YahooMailNeo@web162004.mail.bf1.yahoo.com> Message-ID: Hi Fernando! Thank you for your answer, however I'm trying to set this variable dinamically using a LUA script. Otherwise I have to create a user XML with those preferencies. Thanks Max Il giorno 03/nov/2014, alle ore 19:23, FERNANDO VILLARROEL ha scritto: > I define nobal_amt variable for each user like this: > > > > > > > > > > > > > > > ...... > > > > > So in my dialplan i have: > > > > > > Regards > > > > On Monday, November 3, 2014 2:48 PM, Massimo Varriale wrote: > > > Yes, that's true. > My XML parameter is using just this parameters, however I'm trying to find a way to bill a call based on a rate and reduce the credit accordly. > > In my Lua script I'm trying to use this syntax to build the bridge string: > > > > So, no way to set this variable dinamically? > > Thank you > Max > > > > > > > > Il giorno 03/nov/2014, alle ore 18:35, Avi Marcus ha scritto: > >> The old Wiki says it's a nibblebill configuration variable: >> >> In your conf/autoload_configs/nibblebill.conf.xml add something like this: >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/f7c32b64/attachment.html From msc at freeswitch.org Fri Nov 7 18:45:10 2014 From: msc at freeswitch.org (Michael Collins) Date: Fri, 7 Nov 2014 07:45:10 -0800 Subject: [Freeswitch-users] how to call asterisk extension from freeswitch using mod_event_socket. In-Reply-To: References: Message-ID: The bridge app requires an existing call leg, i.e. the A leg, to connect to the new leg, i.e. the B leg. Also, the bridge app is just that - a dialplan application. It sounds like you need to use an API, that is, a command that you can use at fs_cli. Look up the "originate" command on the wiki and I think you'll be in business. -MC On Fri, Nov 7, 2014 at 6:48 AM, Aqs Younas wrote: > I want to call asterisk extension from freeswitch using ESL. > Here is my python file. > > #!/usr/bin/env python > > import string > import sys > > from ESL import * > > con = ESLconnection("localhost","8021","ClueCon") > #are we connected? > > if con.connected: > con.execute("bridge","sofia/external/555 at 192.168.1.44"); > > When a run this it does not generate call over asterisk extension. > But when i do this using dialplan it works.Here is my default.xml > > > > > > > > > > > > > *But i want to do this using mod_event_socket.* > > Your help would be much appreciated. > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/df8fd597/attachment-0001.html From karl-theo_hofer at inteli-sim.com Fri Nov 7 18:52:20 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Fri, 07 Nov 2014 16:52:20 +0100 Subject: [Freeswitch-users] FS as callback server pass through ring back tone form B leg to A Leg In-Reply-To: <54493C11.8050903@inteli-sim.com> References: <54419311.30902@inteli-sim.com> <5446D2E8.1040407@inteli-sim.com> <5448C5AC.2090302@inteli-sim.com> <54493C11.8050903@inteli-sim.com> Message-ID: <545CEAB4.7000108@inteli-sim.com> Hi Micheal did you have to look at the file i posted to paste bin? With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer at inteli-sim.com kthofer skrev 2014-10-23 19:34: > Hi Michael > https://pastebin.freeswitch.org/23468 > thats the link, to my posting in pastebin. > > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-23 16:50: >> I must have missed the link. I don't see a pb link anywhere here. Can >> you >> re-post? >> -MC >> >> On Thu, Oct 23, 2014 at 2:09 AM, kthofer >> >> wrote: >> >>> Hi Michael >>> >>> do you have some news for me, did you had a look at the pastebin? >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E: karl-theo_hofer at inteli-sim.com >>> >>> kthofer skrev 2014-10-21 23:40: >>> >>> Hi Michael >>> >>> I get a 183 SDP >>> from the b-leg >>> have pated the whole thing >>> debug and sip trace please let me know if you see something that can >>> help. >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> Michael Collins skrev 2014-10-21 01:35: >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? >>> Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> >>> >>> wrote: >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E: karl-theo_hofer at inteli-sim.com >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? >>> Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> >> >>> > >>> wrote: >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>> welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB >>> Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read >>> through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E: karl-theo_hofer at inteli-sim.com >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> Hi Michael >>> >>> I get a 183 SDP >>> from the b-leg >>> have pated the whole thing >>> debug and sip trace please let me know if you see something that can >>> help. >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> >>> Michael Collins skrev 2014-10-21 01:35: >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? >>> Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> >>> wrote: >>> >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E: karl-theo_hofer at inteli-sim.com >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting >>> Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH >>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailing >>> listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe >>> you could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> >>> wrote: >>> >>>> Hi There >>>> >>>> we have FS as a Callback server right now we play our own RB tone >>>> $con->execute("set", "transfer_ringback >>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>>> message. >>>> But that's not really how we like to have it. >>>> instead of our own RB tone we like to pass through the B-leg RB >>>> Tone to >>>> the A-leg. >>>> I know this is a question asked before and we tried to read through >>>> the >>>> forum. >>>> But all suggestions we found and tried did not work. >>>> WE can not make the Box passing through the RB tone from the b-leg >>>> >>>> Any suggestion or how to is highly appreciated. >>>> Hope to get some good answers >>>> >>>> >>>> -- >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> M: +46 7030 22178 >>>> E: karl-theo_hofer at inteli-sim.com >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH >>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailing >>> listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH >>> Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailing >>> listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> I must have missed the link. I don't see a pb link anywhere here. >>> Can you re-post? >>> -MC >>> >>> On Thu, Oct 23, 2014 at 2:09 AM, kthofer >>> >> > wrote: >>> >>> Hi Michael >>> >>> do you have some news for me, did you had a look at the pastebin? >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> >>> kthofer skrev 2014-10-21 23:40: >>>> Hi Michael >>>> >>>> I get a 183 SDP >>>> from the b-leg >>>> have pated the whole thing >>>> debug and sip trace please let me know if you see something >>>> that can help. >>>> >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> Michael Collins skrev 2014-10-21 01:35: >>>>> Are you actually receiving a 183 from the b-leg? Or a 180 even? >>>>> Maybe you >>>>> could pastebin a complete log of a call including SIP trace. >>>>> -MC >>>>> >>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>> >>>>> >>>>> wrote: >>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own RB >>>>>> tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>>>>> welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the B-leg >>>>>> RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read >>>>>> through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from the >>>>>> b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>>> even? Maybe you could pastebin a complete log of a call >>>>>> including SIP trace. >>>>>> -MC >>>>>> >>>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>>> >>>>> >>>>>> >>>>>> > wrote: >>>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own >>>>>> RB tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>>>>> welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the >>>>>> B-leg RB >>>>>> Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read >>>>>> through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from >>>>>> the b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> Hi Michael >>>> >>>> I get a 183 SDP >>>> from the b-leg >>>> have pated the whole thing >>>> debug and sip trace please let me know if you see something >>>> that can help. >>>> >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> Michael Collins skrev 2014-10-21 01:35: >>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>> even? Maybe you >>>>> could pastebin a complete log of a call including SIP trace. >>>>> -MC >>>>> >>>>> On Fri, Oct 17, 2014 at 3:07 PM, >>>>> kthofer >>>>> >>>>> wrote: >>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own RB >>>>>> tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>>>>> welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the B-leg >>>>>> RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read >>>>>> through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from the >>>>>> b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E:karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>>> even? Maybe you could pastebin a complete log of a call >>>>>> including SIP trace. >>>>>> -MC >>>>>> >>>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>>> >>>>> > wrote: >>>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own >>>>>> RB tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a >>>>>> short welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the >>>>>> B-leg RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to >>>>>> read through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from >>>>>> the b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org > > > > Hi Michael > https://pastebin.freeswitch.org/23468 > thats the link, to my posting in pastebin. > > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-23 16:50: >> I must have missed the link. I don't see a pb link anywhere here. Can you >> re-post? >> -MC >> >> On Thu, Oct 23, 2014 at 2:09 AM, kthofer >> wrote: >> >>> Hi Michael >>> >>> do you have some news for me, did you had a look at the pastebin? >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> kthofer skrev 2014-10-21 23:40: >>> >>> Hi Michael >>> >>> I get a 183 SDP >>> from the b-leg >>> have pated the whole thing >>> debug and sip trace please let me know if you see something that can >>> help. >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> Michael Collins skrev 2014-10-21 01:35: >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> >>> wrote: >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >> > >>> wrote: >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB >>> Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read >>> through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> Hi Michael >>> >>> I get a 183 SDP >>> from the b-leg >>> have pated the whole thing >>> debug and sip trace please let me know if you see something that can >>> help. >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> >>> Michael Collins skrev 2014-10-21 01:35: >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >>> could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> wrote: >>> >>> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH ConsultingServices:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailinglistFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe >>> you could pastebin a complete log of a call including SIP trace. >>> -MC >>> >>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>> wrote: >>> >>>> Hi There >>>> >>>> we have FS as a Callback server right now we play our own RB tone >>>> $con->execute("set", "transfer_ringback >>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>>> message. >>>> But that's not really how we like to have it. >>>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>>> the A-leg. >>>> I know this is a question asked before and we tried to read through the >>>> forum. >>>> But all suggestions we found and tried did not work. >>>> WE can not make the Box passing through the RB tone from the b-leg >>>> >>>> Any suggestion or how to is highly appreciated. >>>> Hope to get some good answers >>>> >>>> >>>> -- >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> M: +46 7030 22178 >>>> E:karl-theo_hofer at inteli-sim.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailinglistFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >>> >>> FreeSWITCH-users mailinglistFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> I must have missed the link. I don't see a pb link anywhere here. >>> Can you re-post? >>> -MC >>> >>> On Thu, Oct 23, 2014 at 2:09 AM, kthofer >>> >> > wrote: >>> >>> Hi Michael >>> >>> do you have some news for me, did you had a look at the pastebin? >>> >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E:karl-theo_hofer at inteli-sim.com >>> >>> kthofer skrev 2014-10-21 23:40: >>>> Hi Michael >>>> >>>> I get a 183 SDP >>>> from the b-leg >>>> have pated the whole thing >>>> debug and sip trace please let me know if you see something >>>> that can help. >>>> >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> Michael Collins skrev 2014-10-21 01:35: >>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>> even? Maybe you >>>>> could pastebin a complete log of a call including SIP trace. >>>>> -MC >>>>> >>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>> >>>>> >>>>> wrote: >>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own RB >>>>>> tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>>>>> welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the B-leg >>>>>> RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read >>>>>> through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from the >>>>>> b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>>> even? Maybe you could pastebin a complete log of a call >>>>>> including SIP trace. >>>>>> -MC >>>>>> >>>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>>> >>>>> >>>>>> >>>>>> > wrote: >>>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own >>>>>> RB tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short >>>>>> welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the >>>>>> B-leg RB >>>>>> Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read >>>>>> through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from >>>>>> the b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> Hi Michael >>>> >>>> I get a 183 SDP >>>> from the b-leg >>>> have pated the whole thing >>>> debug and sip trace please let me know if you see something >>>> that can help. >>>> >>>> With best regards >>>> >>>> Karl Theo Hofer >>>> >>>> Michael Collins skrev 2014-10-21 01:35: >>>>> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >>>>> could pastebin a complete log of a call including SIP trace. >>>>> -MC >>>>> >>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>> wrote: >>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own RB tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to read through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from the b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E:karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> Are you actually receiving a 183 from the b-leg? Or a 180 >>>>>> even? Maybe you could pastebin a complete log of a call >>>>>> including SIP trace. >>>>>> -MC >>>>>> >>>>>> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >>>>>> >>>>> > wrote: >>>>>> >>>>>> Hi There >>>>>> >>>>>> we have FS as a Callback server right now we play our own >>>>>> RB tone >>>>>> $con->execute("set", "transfer_ringback >>>>>> \%(400,200,400,450);\%(400,2000,400,450)")right after a >>>>>> short welcome >>>>>> message. >>>>>> But that's not really how we like to have it. >>>>>> instead of our own RB tone we like to pass through the >>>>>> B-leg RB Tone to >>>>>> the A-leg. >>>>>> I know this is a question asked before and we tried to >>>>>> read through the >>>>>> forum. >>>>>> But all suggestions we found and tried did not work. >>>>>> WE can not make the Box passing through the RB tone from >>>>>> the b-leg >>>>>> >>>>>> Any suggestion or how to is highly appreciated. >>>>>> Hope to get some good answers >>>>>> >>>>>> >>>>>> -- >>>>>> With best regards >>>>>> >>>>>> Karl Theo Hofer >>>>>> >>>>>> M: +46 7030 22178 >>>>>> E: karl-theo_hofer at inteli-sim.com >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/39b6ed7f/attachment-0001.html From aqsyounas at gmail.com Fri Nov 7 19:24:27 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 7 Nov 2014 21:24:27 +0500 Subject: [Freeswitch-users] how to call asterisk extension from freeswitch using mod_event_socket. In-Reply-To: References: Message-ID: Thanks for your reply. Actually i figured in out using API with originate. On 7 November 2014 20:45, Michael Collins wrote: > The bridge app requires an existing call leg, i.e. the A leg, to connect > to the new leg, i.e. the B leg. Also, the bridge app is just that - a > dialplan application. It sounds like you need to use an API, that is, a > command that you can use at fs_cli. Look up the "originate" command on the > wiki and I think you'll be in business. > > -MC > > On Fri, Nov 7, 2014 at 6:48 AM, Aqs Younas wrote: > >> I want to call asterisk extension from freeswitch using ESL. >> Here is my python file. >> >> #!/usr/bin/env python >> >> import string >> import sys >> >> from ESL import * >> >> con = ESLconnection("localhost","8021","ClueCon") >> #are we connected? >> >> if con.connected: >> con.execute("bridge","sofia/external/555 at 192.168.1.44"); >> >> When a run this it does not generate call over asterisk extension. >> But when i do this using dialplan it works.Here is my default.xml >> >> >> >> >> >> >> >> >> >> >> >> >> *But i want to do this using mod_event_socket.* >> >> Your help would be much appreciated. >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/aa0b490c/attachment.html From aviv at sent.com Fri Nov 7 19:18:23 2014 From: aviv at sent.com (Aviv Shaham) Date: Fri, 07 Nov 2014 09:18:23 -0700 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415396248535.99205@nexxuspg.com> References: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> <1415396248535.99205@nexxuspg.com> Message-ID: <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555 at your-server.com and then have a dialplan such as: The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. Let me know if you are having any other issues. Aviv On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote: > Hi, > > Thanks for your suggestion, I have tried it and I am able to do a > Inbound call via Nexmo now. But still I am not able to make any > outbound call from my application. > > I have checked the FreeSwitch log by siptrace enable and found that my > call was terminated with a SIP message as " > *IP/2.0 407 Proxy Authentication Required*". > > Please find the siptrace log for my call as an attachment. and let me > know what changes or configuration I need to make for Proxy > Authentication Header. > > Thanks, > > Regards, > Manish Talwar > > > *From:* freeswitch-users-bounces at lists.freeswitch.org > on behalf of Aviv > Shaham *Sent:* 06 November 2014 14:39 *To:* > freeswitch-users at lists.freeswitch.org *Subject:* Re: > [Freeswitch-users] SIP trunking with Nexmo > > Hi Manish, > > Nexmo expects your API KEY to be in the From header. To set the caller > ID you will need to use "caller-id-name". Good timing btw, I just > posted a reply to a similar question on Quora. Have a look: > http://qr.ae/DEbk2 - also covers Plivo. > > Aviv > > > On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: >> Hi, >> >> I have make a SIP Trunking (gateway) in FreeSwitch for connecting >> Nexmo via bridge. I have added this Nexmo file under >> "*\FreeSWITCH\conf\sip_profiles\external*" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: >> >> >> Name Type Data State >> ================================================================================================ >> external-ipv6 profile >> sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) >> 192.168.1.140 alias internal ALIASED external profile >> sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com >> gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway >> sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED >> internal-ipv6 profile >> sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) >> internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) >> ================================================================================================ >> 4 profiles 1 alias >> >> But when I send the request to FreeSwitch by Dial command as: >> *> type="xml/freeswitch-httapi">> applicati**on="set" data="sip_h_api_key=b9c280dd" />> application="set" data="sip_h_**api_secret=7678b8c4" />> application="set" data="sip_h_to=919818753995" /**>> *caller**-id-number="18188535351" context="default" Dialplan="XML" >> >919818753995* >> >> >> here, *18188535351* = Nexmo virtual number for connecting call. >> *919818753995* = mobile number where I am looking for making a call. >> >> It will not connected to Nexmo and call will be terminated with >> message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 >> Originate Failed. Cause: NORMAL_UNSPECIFIED >> >> Please find the FreeSwitch call Log and Nexmo Gateway (which I have >> added in freeswitch conf external folder) as an attachment. >> >> Please let me know whether I am doing SIP trunking in correct way or >> need to change something. >> >> Also, Please suggest me what will be my next step for making a call >> on mobile by this ways. >> >> Thanks, >> >> Regards, Manish Talwar >> >> ___________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites http://www.freeswitch.org >> http://confluence.freeswitch.org http://www.cluecon.com >> >> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org Email had 2 attachments: >> * FsCall.txt >> 15k (text/plain) >> * Nexmo.xml >> 3k (text/xml) > > ___________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites http://www.freeswitch.org > http://confluence.freeswitch.org http://www.cluecon.com > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Email had 1 attachment: > * SipTrace.txt 9k (text/plain) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/210987d0/attachment-0001.html From msc at freeswitch.org Fri Nov 7 20:37:59 2014 From: msc at freeswitch.org (Michael Collins) Date: Fri, 7 Nov 2014 09:37:59 -0800 Subject: [Freeswitch-users] FS as callback server pass through ring back tone form B leg to A Leg In-Reply-To: <545CEAB4.7000108@inteli-sim.com> References: <54419311.30902@inteli-sim.com> <5446D2E8.1040407@inteli-sim.com> <5448C5AC.2090302@inteli-sim.com> <54493C11.8050903@inteli-sim.com> <545CEAB4.7000108@inteli-sim.com> Message-ID: I'll have to defer to those with more experience than I in this scenario. It looks like you receive a 183 and then a 180. I believe there are things you can do to handle this scenario but I must confess I don't know them off the top of my head. -MC On Fri, Nov 7, 2014 at 7:52 AM, kthofer wrote: > Hi Micheal > > did you have to look at the file i posted to paste bin? > > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > kthofer skrev 2014-10-23 19:34: > > Hi Michael > https://pastebin.freeswitch.org/23468 > thats the link, to my posting in pastebin. > > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-23 16:50: > > I must have missed the link. I don't see a pb link anywhere here. Can you > re-post? > -MC > > On Thu, Oct 23, 2014 at 2:09 AM, kthofer > > wrote: > > Hi Michael > > do you have some news for me, did you had a look at the pastebin? > > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > kthofer skrev 2014-10-21 23:40: > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something that can > help. > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB > Tone to > the A-leg. > I know this is a question asked before and we tried to read > through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something that can > help. > > With best regards > > Karl Theo Hofer > > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > wrote: > > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting > Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp:// > www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp:// > www.cudatel.com > > FreeSWITCH-users mailing > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe > you could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp:// > www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp:// > www.cudatel.com > > FreeSWITCH-users mailing > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH > Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp:// > www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp:// > www.cudatel.com > > FreeSWITCH-users mailing > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > I must have missed the link. I don't see a pb link anywhere here. Can you > re-post? > -MC > > On Thu, Oct 23, 2014 at 2:09 AM, kthofer > > wrote: > > Hi Michael > > do you have some news for me, did you had a look at the pastebin? > > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E:karl-theo_hofer at inteli-sim.com > > > kthofer skrev 2014-10-21 23:40: > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something > that can help. > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? > Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short > welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg > RB Tone to > the A-leg. > I know this is a question asked before and we tried to read > through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the > b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 > even? Maybe you could pastebin a complete log of a call > including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > > > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own > RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short > welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the > B-leg RB > Tone to > the A-leg. > I know this is a question asked before and we tried to read > through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from > the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something > that can help. > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe > you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone > to > the A-leg. > I know this is a question asked before and we tried to read through > the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E:karl-theo_hofer at inteli-sim.com > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 > even? Maybe you could pastebin a complete log of a call > including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own > RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a > short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the > B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to > read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from > the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > Hi Michael > https://pastebin.freeswitch.org/23468 > thats the link, to my posting in pastebin. > > > With best regards > > Karl Theo Hofer > > > Michael Collins skrev 2014-10-23 16:50: > > I must have missed the link. I don't see a pb link anywhere here. Can you > re-post? > -MC > > On Thu, Oct 23, 2014 at 2:09 AM, kthofer > wrote: > > > Hi Michael > > do you have some news for me, did you had a look at the pastebin? > > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > kthofer skrev 2014-10-21 23:40: > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something that can > help. > > With best regards > > Karl Theo Hofer > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > wrote: > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB > Tone to > the A-leg. > I know this is a question asked before and we tried to read > through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > Hi Michael > > I get a 183 SDP > from the b-leg > have pated the whole thing > debug and sip trace please let me know if you see something that can > help. > > With best regards > > Karl Theo Hofer > > > Michael Collins skrev 2014-10-21 01:35: > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you > could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > wrote: > > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe > you could pastebin a complete log of a call including SIP trace. > -MC > > On Fri, Oct 17, 2014 at 3:07 PM, kthofer > wrote: > > > Hi There > > we have FS as a Callback server right now we play our own RB tone > $con->execute("set", "transfer_ringback > \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome > message. > But that's not really how we like to have it. > instead of our own RB tone we like to pass through the B-leg RB Tone to > the A-leg. > I know this is a question asked before and we tried to read through the > forum. > But all suggestions we found and tried did not work. > WE can not make the Box passing through the RB tone from the b-leg > > Any suggestion or how to is highly appreciated. > Hope to get some good answers > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > I must have missed the link. I don't see a pb link anywhere here. Can > you re-post? > -MC > > On Thu, Oct 23, 2014 at 2:09 AM, kthofer > wrote: > >> Hi Michael >> >> do you have some news for me, did you had a look at the pastebin? >> >> With best regards >> >> Karl Theo Hofer >> >> M: +46 7030 22178 >> E: karl-theo_hofer at inteli-sim.com >> >> kthofer skrev 2014-10-21 23:40: >> >> Hi Michael >> >> I get a 183 SDP >> from the b-leg >> have pated the whole thing >> debug and sip trace please let me know if you see something that can >> help. >> >> With best regards >> >> Karl Theo Hofer >> >> Michael Collins skrev 2014-10-21 01:35: >> >> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >> could pastebin a complete log of a call including SIP trace. >> -MC >> >> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >> >> wrote: >> >> Hi There >> >> we have FS as a Callback server right now we play our own RB tone >> $con->execute("set", "transfer_ringback >> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >> message. >> But that's not really how we like to have it. >> instead of our own RB tone we like to pass through the B-leg RB Tone to >> the A-leg. >> I know this is a question asked before and we tried to read through the >> forum. >> But all suggestions we found and tried did not work. >> WE can not make the Box passing through the RB tone from the b-leg >> >> Any suggestion or how to is highly appreciated. >> Hope to get some good answers >> >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> M: +46 7030 22178 >> E: karl-theo_hofer at inteli-sim.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >> could pastebin a complete log of a call including SIP trace. >> -MC >> >> On Fri, Oct 17, 2014 at 3:07 PM, kthofer > > >> wrote: >> >> Hi There >> >> we have FS as a Callback server right now we play our own RB tone >> $con->execute("set", "transfer_ringback >> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >> message. >> But that's not really how we like to have it. >> instead of our own RB tone we like to pass through the B-leg RB >> Tone to >> the A-leg. >> I know this is a question asked before and we tried to read >> through the >> forum. >> But all suggestions we found and tried did not work. >> WE can not make the Box passing through the RB tone from the b-leg >> >> Any suggestion or how to is highly appreciated. >> Hope to get some good answers >> >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> M: +46 7030 22178 >> E: karl-theo_hofer at inteli-sim.com >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> Hi Michael >> >> I get a 183 SDP >> from the b-leg >> have pated the whole thing >> debug and sip trace please let me know if you see something that can >> help. >> >> With best regards >> >> Karl Theo Hofer >> >> >> Michael Collins skrev 2014-10-21 01:35: >> >> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe you >> could pastebin a complete log of a call including SIP trace. >> -MC >> >> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >> wrote: >> >> >> Hi There >> >> we have FS as a Callback server right now we play our own RB tone >> $con->execute("set", "transfer_ringback >> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >> message. >> But that's not really how we like to have it. >> instead of our own RB tone we like to pass through the B-leg RB Tone to >> the A-leg. >> I know this is a question asked before and we tried to read through the >> forum. >> But all suggestions we found and tried did not work. >> WE can not make the Box passing through the RB tone from the b-leg >> >> Any suggestion or how to is highly appreciated. >> Hope to get some good answers >> >> >> -- >> With best regards >> >> Karl Theo Hofer >> >> M: +46 7030 22178 >> E: karl-theo_hofer at inteli-sim.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> Are you actually receiving a 183 from the b-leg? Or a 180 even? Maybe >> you could pastebin a complete log of a call including SIP trace. >> -MC >> >> On Fri, Oct 17, 2014 at 3:07 PM, kthofer >> wrote: >> >>> Hi There >>> >>> we have FS as a Callback server right now we play our own RB tone >>> $con->execute("set", "transfer_ringback >>> \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome >>> message. >>> But that's not really how we like to have it. >>> instead of our own RB tone we like to pass through the B-leg RB Tone to >>> the A-leg. >>> I know this is a question asked before and we tried to read through the >>> forum. >>> But all suggestions we found and tried did not work. >>> WE can not make the Box passing through the RB tone from the b-leg >>> >>> Any suggestion or how to is highly appreciated. >>> Hope to get some good answers >>> >>> >>> -- >>> With best regards >>> >>> Karl Theo Hofer >>> >>> M: +46 7030 22178 >>> E: karl-theo_hofer at inteli-sim.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/e2ce3380/attachment-0001.html From jcabezas at inovax.com.br Fri Nov 7 20:52:29 2014 From: jcabezas at inovax.com.br (Julio Cabezas) Date: Fri, 07 Nov 2014 15:52:29 -0200 Subject: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs In-Reply-To: References: <02f201cff9e9$4cd09cf0$e671d6d0$@inovax.com.br> Message-ID: <545D06DD.4020204@inovax.com.br> Thanks for the help, guys ! On 11/7/2014 11:27 AM, Brian West wrote: > https://freeswitch.org/fisheye/changelog/freeswitch?cs=08ff88ec31572071c5e5dbe6fad51578814c4bd9 > > Not hard :) > > This could help. > > On Thu, Nov 6, 2014 at 4:46 PM, Kristian Kielhofner > wrote: > > This won't be as bad as you think it is. Write something to > generate it statically or return the XML via mod_xml_curl and > Sofia will handle it just fine. > > I had to lab up some crazy scenario once where I created hundreds > of gateways on a profile and it worked just fine. > > > On Thursday, November 6, 2014, Julio Cesar Esteves Cabezas > > wrote: > > I have 100+ AORS to register with the main server. > So, as far as I understand, I have to define 100+ gateways to > the same main > SIP server in FS, each one doing a single AOR registration. > Is there any support for multiple AOR registrations per gateway ? > Or any other more "compact" scheme ? > > Thanks. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On > Behalf Of Michael > Jerris > Sent: Thursday, November 06, 2014 11:52 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Registering FreeSWITCH with > another server, > multiples AORs > > The mod_sofia concept that makes outbound registrations is > called a gateway. > > > On Nov 6, 2014, at 7:50 AM, Julio Cabezas > wrote: > > > > Hi, > > How to make FreeSWITCH Sofia UA request several SIP > registrations > > (with different AORs) to an already deployed main SIP server ? > > I need that for that server forward some calls to FS. > > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Sent from mobile device > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > > */Brian West/* > brian at freeswitch.org > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/893de68a/attachment.html From rajgopalfs at gmail.com Fri Nov 7 20:48:34 2014 From: rajgopalfs at gmail.com (raj gopal) Date: Fri, 7 Nov 2014 12:48:34 -0500 Subject: [Freeswitch-users] unable to do "api uuid_getvar" in outbound socket created by originate command Message-ID: I am trying to "api" commands on outbound socket created by originate .... &(socket ..async full) and I am getting -ERR command not found. So I tried same thing on fs_cli and netcat. from fs_cli executed following command: originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 async full) I did a netcat: on 192.168.1.94 9050. when the extension 1000 answered, I did connect\n\n , myevents\n\n and sendmsg command to answer (i.e. answers\n\n). now if I execute api uuid_setvar uuid x 30 or api uuid_getvar uuid x; I am getting -ERR command not found. if I do same stuff on outbound socket created when extension 1000 dials in to extension 1021. and I setup dialplan to created outbound socket application=socket data="192.168.1.94L9050 async full" I could do api commands. What is the difference in both outbound sockets?? except one is created using originate other is created when dialed in. My intension is to dial a phone number, do play_and_get_digits and do get the digits pressed by user using uuid_getvar in outbound socket but the originate command it self is issued in inbound socket. appreciate any insights. Regards, Raj -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/9afad695/attachment.html From msh.computing at gmail.com Sat Nov 8 01:20:41 2014 From: msh.computing at gmail.com (Steve Kieu) Date: Sat, 8 Nov 2014 08:20:41 +1000 Subject: [Freeswitch-users] Newbie question: Set up to answer new phone number (please be gentle) In-Reply-To: <1415285435.32669.YahooMailBasic@web160204.mail.bf1.yahoo.com> References: <1415285435.32669.YahooMailBasic@web160204.mail.bf1.yahoo.com> Message-ID: You better start your FS new installation and start from there, reading the start up guide in the wiki, step by step understand how FS process the call flow, how the profile works and the ACL stuff. In a fresh installation of FS if you just change the default password and then get extension 100[0-9] it should automatically work. In your symptom it looks like call initialted but FS need to auth the call (for all calls coming into internal profile and you want that to protect call fraud). But the caller can not auth or not responding to the auth request. Again, search the wiki and read again and again ... - If you find a proper wiki section but do not understand the wording then post it here we can explain more. On Fri, Nov 7, 2014 at 12:50 AM, T Fred Farmington wrote: > Since I am a total newbie to FreeSwitch please be patient and 'gentle' > with me.... > > I have 'inherited' 2 FreeSwitch installations. One is working (version > 1.0.7) and one (version 1.5.14) is a new test environment that is not > working. > > I cannot get the new test environment to answer the inbound call and in > the FreeSwitch log I find: Rejected by acl "domains" > > I have tried to search the FreeSwitch documentation for an answer and got > lost in it. > And I have tried to search the web for an answer, but did not find > anything clearly spelled out. > And, lastly, I tried to investigate the working installation, but could > not find my answer. > > I have a dedicated SIP phone line which can call into the new test > environment. > And via FS_CLI I can see the call being received. > However the connection is never established and in the log I find: > Rejected by acl "domains" > > In both FreeSwitch environments I have a directory named: \Debug into > which the previous individual installed all of the configuration parameters. > > Questions: > 1. Where/How is FreeSwitch configured to recognize this directory and look > for its parameters there? > **** Maybe the non-working environment does not have that setting > established and therefore is ignoring settings made there. > > 2. I have configured the new inbound IP address into the file: > acl.conf.xml and have 'mirrored' the working environment in that regard. > **** But the 'Rejected' message in the log suggests that somehow that is > not being recognized. > > 3. Are there other files which I have missed in trying to set this up? > > Thanks > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Steve Kieu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141108/5689dfa0/attachment-0001.html From brian at freeswitch.org Sat Nov 8 02:01:55 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 7 Nov 2014 17:01:55 -0600 Subject: [Freeswitch-users] Newbie question: Set up to answer new phone number (please be gentle) In-Reply-To: References: <1415285435.32669.YahooMailBasic@web160204.mail.bf1.yahoo.com> Message-ID: Sounds like your client is getting a 401 by FreeSWITCH and never actually receiving it, 'sofia global siptrace on' you'll probably see a 401 go out and the device won't answer the challenge because it never receives it, the rejected by domains acl thing isn't the problem usually its misunderstanding about what is expected vs what is actually taking place. On Fri, Nov 7, 2014 at 4:20 PM, Steve Kieu wrote: > You better start your FS new installation and start from there, reading > the start up guide in the wiki, step by step understand how FS process the > call flow, how the profile works and the ACL stuff. > > In a fresh installation of FS if you just change the default password and > then get extension 100[0-9] it should automatically work. > > In your symptom it looks like call initialted but FS need to auth the call > (for all calls coming into internal profile and you want that to protect > call fraud). But the caller can not auth or not responding to the auth > request. > > Again, search the wiki and read again and again ... - If you find a proper > wiki section but do not understand the wording then post it here we can > explain more. > > > > On Fri, Nov 7, 2014 at 12:50 AM, T Fred Farmington > wrote: > >> Since I am a total newbie to FreeSwitch please be patient and 'gentle' >> with me.... >> >> I have 'inherited' 2 FreeSwitch installations. One is working (version >> 1.0.7) and one (version 1.5.14) is a new test environment that is not >> working. >> >> I cannot get the new test environment to answer the inbound call and in >> the FreeSwitch log I find: Rejected by acl "domains" >> >> I have tried to search the FreeSwitch documentation for an answer and got >> lost in it. >> And I have tried to search the web for an answer, but did not find >> anything clearly spelled out. >> And, lastly, I tried to investigate the working installation, but could >> not find my answer. >> >> I have a dedicated SIP phone line which can call into the new test >> environment. >> And via FS_CLI I can see the call being received. >> However the connection is never established and in the log I find: >> Rejected by acl "domains" >> >> In both FreeSwitch environments I have a directory named: \Debug into >> which the previous individual installed all of the configuration parameters. >> >> Questions: >> 1. Where/How is FreeSwitch configured to recognize this directory and >> look for its parameters there? >> **** Maybe the non-working environment does not have that setting >> established and therefore is ignoring settings made there. >> >> 2. I have configured the new inbound IP address into the file: >> acl.conf.xml and have 'mirrored' the working environment in that regard. >> **** But the 'Rejected' message in the log suggests that somehow that is >> not being recognized. >> >> 3. Are there other files which I have missed in trying to set this up? >> >> Thanks >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Steve Kieu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141107/2610a88c/attachment.html From jerry.richards at teotech.com Sat Nov 8 03:22:22 2014 From: jerry.richards at teotech.com (Jerry Richards) Date: Sat, 8 Nov 2014 00:22:22 +0000 Subject: [Freeswitch-users] Cannot Establish Audio Call First, Then Enable Video Message-ID: <4720b62aeb15451dad4d5f52514e6177@BN1PR04MB405.namprd04.prod.outlook.com> Using two Bria softphones, if I establish an audio call first, and then turn on video, the softphone sends a re-INVITE including the H264 codec, but Freeswitch does not forward the INVITE to the other endpoint. If I originate the call as a video call, then the video call is established successfully. I tried this with the latest Freeswitch as of yesterday. The only change I made to the default configuration was to add the H264 codec to and in vars.xml. Is there a configuration tag that would allow this? Or is it required that the video codec must be offered in the first INVITE for video to work (i.e. cannot be added later)? Regards, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141108/0442b47e/attachment.html From krice at freeswitch.org Sat Nov 8 04:27:10 2014 From: krice at freeswitch.org (Ken Rice) Date: Sat, 08 Nov 2014 01:27:10 +0000 Subject: [Freeswitch-users] FreeSWITCH founder gets movie credit Message-ID: <545d716e74c28_47676af32483244@ip-10-142-130-179.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/1ACmdcc FreeSWITCH founder gets movie credit I?m an extra in a movie but I?m not sure you ever see me. My son has a real part though as a kid named Sam and ends up in a sticky situation to say the least. Check out Available on iTunes and DVD! -Anthony Minessale -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141108/ef67ef32/attachment.html From manish.talwar at nexxuspg.com Sat Nov 8 09:10:00 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Sat, 8 Nov 2014 06:10:00 +0000 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> References: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> <1415396248535.99205@nexxuspg.com>, <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> Message-ID: <1415475715878.91825@nexxuspg.com> Hi, Thanks for your suggestion, I have make these changes and removed the L16 codec from request now. I have set "absolute_codec_string" and "nexmo_forwarded_for" and its not throwing any error message in SIP trace now. But still, I am not able to make a call on my mobile number "1919818753995". Its show message on FreeSwitch log as "[RECOVERY_ON_TIMER_EXPIRE]" and hangup the freeswitch call. Also, there is no log created on Nexmo dashboard for this call's. I am sending my call request to Nexmo from FreeSwitch by dialplan as. Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" ;tag=D8g4a5NvH4emF To: I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham Sent: 07 November 2014 21:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555 at your-server.com and then have a dialplan such as: The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. Let me know if you are having any other issues. Aviv On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote: Hi, Thanks for your suggestion, I have tried it and I am able to do a Inbound call via Nexmo now. But still I am not able to make any outbound call from my application. I have checked the FreeSwitch log by siptrace enable and found that my call was terminated with a SIP message as " IP/2.0 407 Proxy Authentication Required". Please find the siptrace log for my call as an attachment. and let me know what changes or configuration I need to make for Proxy Authentication Header. Thanks, Regards, Manish Talwar ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 06 November 2014 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo. Aviv On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: Hi, I have make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: 919818753995 here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain) -------------- next part -------------- An HTML attachment was scrubbed... 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Name: SipTrace.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141108/5ac08513/attachment-0001.txt From rajgopalfs at gmail.com Sat Nov 8 20:23:01 2014 From: rajgopalfs at gmail.com (raj) Date: Sat, 8 Nov 2014 17:23:01 +0000 (UTC) Subject: [Freeswitch-users] unable to do References: Message-ID: > > I am trying to "api" commands on outbound socket created by originate .... &(socket ..async full) and I am getting -ERR command not found.So I tried same thing on fs_cli and netcat. > > from fs_cli executed following command: > > originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 async full) > > I did a netcat: on 192.168.1.94 9050. > > when the extension 1000 answered, I did connect\n\n , myevents\n\n and sendmsg command to answer (i.e. ?answers\n\n). > > now if I execute api uuid_setvar uuid x 30 or ?api uuid_getvar uuid x;? > I am getting -ERR command not found. > > if I do same stuff on outbound socket created when extension 1000 dials in to extension 1021. > and I setup dialplan to created outbound socket > application=socket ?data="192.168.1.94L9050 async full" ?I could do api commands. What is the difference in both outbound sockets?? except one is created using originate other is created when dialed in.? > My intension is to dial a phone number, do play_and_get_digits and do get the digits pressed by user using uuid_getvar in outbound socket but the originate command it self is issued in inbound socket.? > > appreciate any insights. > Regards, > Raj > > > raj gopal writes: I did some debugging mod_event_socket.c apparently it it not getting paramaters "async full" when given in originate command. It appears to be a bug. I am using 1.4.13. can somebody confirm this?? Thanks. ________________________________________________________________________ _ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users > http://www.freeswitch.org From bk1379 at yahoo.com Sun Nov 9 07:50:19 2014 From: bk1379 at yahoo.com (babak) Date: Sat, 8 Nov 2014 20:50:19 -0800 Subject: [Freeswitch-users] Freeswitch as load balancer in media proxy mode Message-ID: <1415508619.56507.YahooMailNeo@web141501.mail.bf1.yahoo.com> Hi I want to use a freeswitch in media proxy mode as load balancer in front of two other freeswitch servers. Is there any sample configuration guide? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141108/db91f426/attachment.html From info at snelgoedkoop.nl Sun Nov 9 16:44:45 2014 From: info at snelgoedkoop.nl (Martin Snelgoedkoop) Date: Sun, 09 Nov 2014 14:44:45 +0100 Subject: [Freeswitch-users] Extension and carrier same ip problem Message-ID: <545F6FCD.4050000@snelgoedkoop.nl> > Hi, we use Kazoo which loads acl and dial plan out a DB. > We run in to smth that seems freeswitch default behavior so that's why I write here. > > We have a carrier that needs to register a extension to our platform as well. > They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct. > > The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route. > > I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy. > > The setup is kamailio>freeswitch > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/3e0d7370/attachment.html From ifink at machshevet.com Sun Nov 9 13:33:47 2014 From: ifink at machshevet.com (=?utf-8?Q?Israel_Fink?=) Date: Sun, 9 Nov 2014 10:33:47 +0000 Subject: [Freeswitch-users] =?utf-8?q?_Re=3A__Confusing_about_FreeSWITCH_c?= =?utf-8?q?allID_variables?= In-Reply-To: References: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com>, Message-ID: <545f8b31.a96db40a.58b0.3c2b@mx.google.com> Thanks for your clear answer. But I find thinks a bit different. we do collect the FS events during ESL mod event_socket, and find that the Unique-ID changes by channel. What I try to do is, we have a FS server, each call comes in via the server, there were made some logic like answer, greeting etc. and also saving the call in DB. from there it goes to a client extension to get the call, speak, hang up etc. when it comes to the client I need to know which call it is in the server and in the DB. The only variable the client gets from the server is sip_call_id, and now after your explanation I understand why, since this is outbound. My problem is that this variable also changes when bridging, it begins with one value then when bridging it changes to another value. the client gets the second value after bridging, and I cannot find this new value in early events like 'CHANNEL_CREATE' or 'CHANNEL_ORIGINATE', it only appears in later events like 'CHANNEL_PROGRESS' when the client already ringing, and this is a bit late. In short, I need a unique variable to can recognize the incoming call to the client with the incoming call to the server. Israel Fink - Developer Machshevet team From: Brian West Sent: ?Friday?, ?November? ?7?, ?2014 ?12?:?23? ?AM To: freeswitch-users at lists.freeswitch.org 1. Unique-ID is the session UUID, this won't change at all during the session (aka call). 2. Channel-Call-UUID is either the same as Unique-ID or the Unique-ID of the session you're bridged to by looking at the source code. 3. sip_call_id will be the sip call_id of an inbound/outbound SIP call, More than likely the outbound would match Unique-ID and on inbound it would be what ever the remote party sets/sends. The bigger question is what exactly are you trying to solve/understand? On Thu, Nov 6, 2014 at 7:52 AM, Israel Fink wrote: I'm a bit confusing about the channel variables that represent the call ID. I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID 3) variable_sip_call_id, what is the different between this. The value of Channel-Call-UUID seems to be stable during the entire call, the value of Unique-ID in the begin it is the same as Channel-Call-UUID but when bridging it changes it value to another ID, then it comes back to the beginning value, i.m not clear when and why. The variable variable_sip_call_id many time don't have any value, and also it changes the value when bridging and then comes back to the previous value. I have looked for an explanation in FreeSWITCH wiki, but don't find. In the mailing list a have found an explanation here: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, but still not enough. Can someone give a good explanation about this variables,what is their purpose, and so on. Israel Fink - Developer Machshevet team _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/c7b4e612/attachment-0001.html From voransoy at gmail.com Sun Nov 9 23:57:48 2014 From: voransoy at gmail.com (Volkan Oransoy) Date: Sun, 9 Nov 2014 22:57:48 +0200 Subject: [Freeswitch-users] Json_cdr vs cdr_mongo Message-ID: Hi all, I am working on a CDR reporting tool and I am testing both mongo cdr and json cdr. With mongo cdr, there is no problem and I am getting all variables as expected. With json cdr, I am sending records to couchdb. I works well with a minor diffrence. Mongo shows all callflow in an array but couchdb (with json cdr), callflow variable shows only first flow element. Am I missing a point or is this a normal behaviour? Thanks, /Volkan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/a9ede7ec/attachment.html From steveayre at gmail.com Mon Nov 10 02:05:42 2014 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 9 Nov 2014 23:05:42 +0000 Subject: [Freeswitch-users] Confusing about FreeSWITCH callID variables In-Reply-To: <545f8b31.a96db40a.58b0.3c2b@mx.google.com> References: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> <545f8b31.a96db40a.58b0.3c2b@mx.google.com> Message-ID: sip_call_id is the Call-ID header of the SIP packet. It will usually be different for the A-leg and B-leg of the call. However once in a call it identifies the call and *must* stay the same. If you are seeing it change then you're either looking at different legs of the call. On 9 November 2014 10:33, Israel Fink wrote: > Thanks for your clear answer. > But I find thinks a bit different. we do collect the FS events during ESL > mod event_socket, and find that the Unique-ID changes by channel. > What I try to do is, we have a FS server, each call comes in via the > server, there were made some logic like answer, greeting etc. and also > saving the call in DB. from there it goes to a client extension to get the > call, speak, hang up etc. when it comes to the client I need to know which > call it is in the server and in the DB. > The only variable the client gets from the server is sip_call_id, and now > after your explanation I understand why, since this is outbound. > My problem is that this variable also changes when bridging, it begins > with one value then when bridging it changes to another value. the client > gets the second value after bridging, and I cannot find this new value in > early events like 'CHANNEL_CREATE' or 'CHANNEL_ORIGINATE', it only appears > in later events like 'CHANNEL_PROGRESS' when the client already ringing, > and this is a bit late. > In short, I need a unique variable to can recognize the incoming call to > the client with the incoming call to the server. > > Israel Fink - Developer > Machshevet team > > *From:* Brian West > *Sent:* ?Friday?, ?November? ?7?, ?2014 ?12?:?23? ?AM > *To:* freeswitch-users at lists.freeswitch.org > > 1. Unique-ID is the session UUID, this won't change at all during the > session (aka call). > > 2. Channel-Call-UUID is either the same as Unique-ID or the Unique-ID of > the session you're bridged to by looking at the source code. > > 3. sip_call_id will be the sip call_id of an inbound/outbound SIP call, > More than likely the outbound would match Unique-ID and on inbound it would > be what ever the remote party sets/sends. > > The bigger question is what exactly are you trying to solve/understand? > > On Thu, Nov 6, 2014 at 7:52 AM, Israel Fink wrote: > >> I'm a bit confusing about the channel variables that represent the call >> ID. >> >> I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID >> 3) variable_sip_call_id, what is the different between this. >> >> The value of Channel-Call-UUID seems to be stable during the entire call, >> the value of Unique-ID in the begin it is the same as Channel-Call-UUID but >> when bridging it changes it value to another ID, then it comes back to the >> beginning value, i.m not clear when and why. >> >> The variable variable_sip_call_id many time don't have any value, and >> also it changes the value when bridging and then comes back to the previous >> value. >> >> I have looked for an explanation in FreeSWITCH wiki, but don't find. >> >> In the mailing list a have found an explanation here: >> http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, >> but still not enough. >> Can someone give a good explanation about this variables,what is their >> purpose, and so on. >> >> >> Israel Fink - Developer >> Machshevet team >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/47c431b4/attachment.html From andretodd at verizon.net Mon Nov 10 03:39:00 2014 From: andretodd at verizon.net (Andre Demattia) Date: Sun, 09 Nov 2014 19:39:00 -0500 Subject: [Freeswitch-users] hash_remote | [name]| Message-ID: <0NES00DROSH88X10@vms173005.mailsrvcs.net> Hi, What would the syntax look like to connect more than one FreeSwitch Limit hash_remote | [name]| Thanks Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/ae8959a3/attachment.html From lfurrea at gmail.com Mon Nov 10 07:14:47 2014 From: lfurrea at gmail.com (Luis F Urrea) Date: Sun, 9 Nov 2014 20:14:47 -0800 Subject: [Freeswitch-users] DTLS-SRTP decryption Message-ID: Hi all, So I have been doing some research on the feasibility of decrypting SRTP traffic whose key has been negotiated through DTLS, such as WebRTC RTP traffic. It seems to me that even when having the private keys from the DTLS negotiation, the SRTP master key is obtained through a TLS-exporter mechanism as defined in RFC5705 , and therefore I haven't found any way to obtain this from a packet capture. It only seems reasonable that an endpoint in the negotiation would provide this. Is there anything that could be done in the FS side of the DTLS negotiation to obtain the SRTP master key? Thanks in advance for your input. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141109/0b54ad08/attachment.html From kalyaneekulkarni at gmail.com Mon Nov 10 08:52:06 2014 From: kalyaneekulkarni at gmail.com (Kalyani Kulkarni) Date: Mon, 10 Nov 2014 11:22:06 +0530 Subject: [Freeswitch-users] unable to do (raj) Message-ID: > > > originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 > async full) > try 192.168.1.94:9050 async full, the colon is missing in your syntax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/17c93da5/attachment-0001.html From GB at cm.nl Mon Nov 10 13:47:25 2014 From: GB at cm.nl (Grant Bagdasarian) Date: Mon, 10 Nov 2014 11:47:25 +0100 Subject: [Freeswitch-users] ESL: Which event is generated when an api command completed? In-Reply-To: References: Message-ID: Is there a way to pass a UUID (Event-UUID) for the application being executed by uuid_broadcast using the format: ?uuid_broadcast app[![hangup_cause]]::args [aleg|bleg|both]? uuid_broadcast a69b7fc0-68c5-11e4-937f-4fc1451152d2 playback::/usr/src/freeswitch/sounds/prompts/welcome.wav both I?m not sure how to format the ?args? when executing playback or any other dialplan application. The Wiki page for playback allows the playback data to contain variables enclosed in curly braces.https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_playback But then I need to listen for PLAYBACK_STOP, which won?t be a problem, but a while back I read on the mailinglist not to use PLAYBACK_START or STOP because it wasn?t reliable? I?m using version 1.2.13. Not sure to which version that applied. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Wednesday, November 5, 2014 10:49 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ESL: Which event is generated when an api command completed? For bgapi it's https://wiki.freeswitch.org/wiki/Event_list#BACKGROUND_JOB On 5 November 2014 09:43, Grant Bagdasarian > wrote: Hello, When using Outbound ESL in async full mode and executing a API command like uuid_transfer or uuid_bridge using bgapi, which channel event is generated to notify the command execution completed? For dialplan applications it is the CHANNEL_EXECUTE_COMPLETED event. That I know. All I got is this: Command Executed: uuid_transfer 563c9750-64cc-11e4-8cd8-4fc1451152d2 -both park_extension XML park_ext Bgapi returned an ESLevent: { "Event-Name": "SOCKET_DATA", "Content-Type": "command/reply", "Reply-Text": "+OK Job-UUID: ede6c869-0bfa-4e07-8490-04973b6ba971", "Job-UUID": "ede6c869-0bfa-4e07-8490-04973b6ba971" } However, a channel event named SOCKET_DATA did not show up when listening for channel events using the RecvEvent method. What I?m trying to do is use bgapi or sendMSG to execute an application/command, and use the RecvEvent method to wait for a channel event that indicates the application/command completed execution. So if I were to use the playback application, I?d send it using the SendMSG method, and use RecvEvent to wait for the CHANNEL_EXECUTE_COMPLETED event containing the UUID set for this playback. This works fine for dialplan applications but I don?t know which event to listen for when using api commands. I hope someone could point me into the right direction. Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/77013654/attachment.html From ifink at machshevet.com Mon Nov 10 14:05:36 2014 From: ifink at machshevet.com (=?utf-8?Q?Israel_Fink?=) Date: Mon, 10 Nov 2014 11:05:36 +0000 Subject: [Freeswitch-users] =?utf-8?q?_Re=3A__Confusing_about_FreeSWITCH_c?= =?utf-8?q?allID_variables?= In-Reply-To: References: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> <545f8b31.a96db40a.58b0.3c2b@mx.google.com>, Message-ID: <54609d28.43c0c20a.1f0e.ffffe78a@mx.google.com> Regards my problem with the client, I already solved the problem in code. To be clear about FS, is there any another variable that is unique for the 'call' no matter which call leg? Israel Fink - Developer Machshevet team From: Steven Ayre Sent: ?Monday?, ?November? ?10?, ?2014 ?1?:?05? ?AM To: freeswitch-users at lists.freeswitch.org sip_call_id is the Call-ID header of the SIP packet. It will usually be different for the A-leg and B-leg of the call. However once in a call it identifies the call and *must* stay the same. If you are seeing it change then you're either looking at different legs of the call. On 9 November 2014 10:33, Israel Fink wrote: Thanks for your clear answer. But I find thinks a bit different. we do collect the FS events during ESL mod event_socket, and find that the Unique-ID changes by channel. What I try to do is, we have a FS server, each call comes in via the server, there were made some logic like answer, greeting etc. and also saving the call in DB. from there it goes to a client extension to get the call, speak, hang up etc. when it comes to the client I need to know which call it is in the server and in the DB. The only variable the client gets from the server is sip_call_id, and now after your explanation I understand why, since this is outbound. My problem is that this variable also changes when bridging, it begins with one value then when bridging it changes to another value. the client gets the second value after bridging, and I cannot find this new value in early events like 'CHANNEL_CREATE' or 'CHANNEL_ORIGINATE', it only appears in later events like 'CHANNEL_PROGRESS' when the client already ringing, and this is a bit late. In short, I need a unique variable to can recognize the incoming call to the client with the incoming call to the server. Israel Fink - Developer Machshevet team From: Brian West Sent: ?Friday?, ?November? ?7?, ?2014 ?12?:?23? ?AM To: freeswitch-users at lists.freeswitch.org 1. Unique-ID is the session UUID, this won't change at all during the session (aka call). 2. Channel-Call-UUID is either the same as Unique-ID or the Unique-ID of the session you're bridged to by looking at the source code. 3. sip_call_id will be the sip call_id of an inbound/outbound SIP call, More than likely the outbound would match Unique-ID and on inbound it would be what ever the remote party sets/sends. The bigger question is what exactly are you trying to solve/understand? On Thu, Nov 6, 2014 at 7:52 AM, Israel Fink wrote: I'm a bit confusing about the channel variables that represent the call ID. I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID 3) variable_sip_call_id, what is the different between this. The value of Channel-Call-UUID seems to be stable during the entire call, the value of Unique-ID in the begin it is the same as Channel-Call-UUID but when bridging it changes it value to another ID, then it comes back to the beginning value, i.m not clear when and why. The variable variable_sip_call_id many time don't have any value, and also it changes the value when bridging and then comes back to the previous value. I have looked for an explanation in FreeSWITCH wiki, but don't find. In the mailing list a have found an explanation here: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, but still not enough. Can someone give a good explanation about this variables,what is their purpose, and so on. Israel Fink - Developer Machshevet team _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/0c8474c1/attachment-0001.html From idokan at gmail.com Mon Nov 10 15:53:57 2014 From: idokan at gmail.com (ik) Date: Mon, 10 Nov 2014 14:53:57 +0200 Subject: [Freeswitch-users] setting recording codec Message-ID: Hello, I could not find any documentation on this subject. I require to have a recording of an a-law wav file (RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). Can I set a variable that tells the record command what type of codec to store the file with ? I know how to convert a normal wav recording into it using ffmpeg (for example): $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav But I wish to avoid it, if I can use the record app, or use the uuid_record API for it. Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/7778d3da/attachment.html From snabel at lexifone.com Mon Nov 10 17:37:07 2014 From: snabel at lexifone.com (Snabel Kabiya) Date: Mon, 10 Nov 2014 16:37:07 +0200 Subject: [Freeswitch-users] execute a lua script from another script Message-ID: Hi, Is there a way to execute a lua script from another script? I've this script s1.lua i want to run s2.lua from it. I've tried s1.lua: dofile(s2.lua); but this didn't work. Thanks, Snabel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/f4cbc94e/attachment.html From balazs.sandor at virtual-call-center.eu Mon Nov 10 18:04:47 2014 From: balazs.sandor at virtual-call-center.eu (=?UTF-8?B?QmFsw6F6cyBTw6FuZG9y?=) Date: Mon, 10 Nov 2014 16:04:47 +0100 Subject: [Freeswitch-users] mode_verto recive dtmf Message-ID: Hi all! I would like to receive dtmf with mod_verto. How can I do that? Best regards, Bal?zs S?NDOR Software Developer Virtual Call Center MUNICH | BUDAPEST | WARSAW Phone: +44 (0) 863 801 69 Web: www.virtual-call-center.eu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/833e3842/attachment.html From dion at openlogic.com.au Mon Nov 10 16:11:36 2014 From: dion at openlogic.com.au (Dion Phillips) Date: Mon, 10 Nov 2014 21:11:36 +0800 Subject: [Freeswitch-users] Freeswitch port numbers assigned to phones Message-ID: <5460B988.4040209@openlogic.com.au> Hi All I am looking for some insight into FS port numbers. The default internal port number is 5060 and when the IP phones register with FS, one of the phones gets assigned 5060 and the remaining phones get a random port number assigned (I am using this command to see the port numbers: "sofia status profile internal reg"). How is this random port number decided on? How can I tie a phone to a specific port number? Thanks Dion. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/c2af134e/attachment.html From sbirla at tampahost.net Mon Nov 10 18:12:55 2014 From: sbirla at tampahost.net (Sumit Birla) Date: Mon, 10 Nov 2014 10:12:55 -0500 Subject: [Freeswitch-users] execute a lua script from another script In-Reply-To: References: Message-ID: <864F3C77-46B1-4000-9D8E-9F2D5BE883D6@tampahost.net> Try using the complete path to s2.lua in quotes. > On Nov 10, 2014, at 9:37 AM, Snabel Kabiya wrote: > > Hi, > > Is there a way to execute a lua script from another script? > I've this script s1.lua i want to run s2.lua from it. I've tried > > > s1.lua: > dofile(s2.lua); > > but this didn't work. > > Thanks, > Snabel > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From alhakeem at gmail.com Mon Nov 10 18:18:58 2014 From: alhakeem at gmail.com (Abdul Hakeem) Date: Mon, 10 Nov 2014 15:18:58 +0000 Subject: [Freeswitch-users] C Dialplan processing routines Message-ID: Hello, Could anyone give me an example on how to link a C code based routine to invoke FS API's to process not just Dialplan or Chatplan but other events such as Register, Reinvite etc ? Thank you in advance. Regards, Abdul Hakeem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/6d56127b/attachment.html From flokrrr at gmail.com Mon Nov 10 18:36:05 2014 From: flokrrr at gmail.com (Florent Krieg) Date: Mon, 10 Nov 2014 16:36:05 +0100 Subject: [Freeswitch-users] execute a lua script from another script In-Reply-To: References: Message-ID: Hello Snabel, Is it within the dialplan? If it's the case you can do: session:execute("lua", "s2.lua $params") Otherwise I don't know. Florent 2014-11-10 15:37 GMT+01:00 Snabel Kabiya : > Hi, > > Is there a way to execute a lua script from another script? > I've this script s1.lua i want to run s2.lua from it. I've tried > > > s1.lua: > dofile(s2.lua); > > but this didn't work. > > Thanks, > Snabel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/ea8275dd/attachment.html From denis at ringme.ru Mon Nov 10 18:50:47 2014 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Mon, 10 Nov 2014 18:50:47 +0300 Subject: [Freeswitch-users] troubles with regex in mod_ivr? Message-ID: <5460DED7.3000702@ringme.ru> We need action on timeout, made digits = "/^$/" - nothing. How to do it correctly? From igorolhovskiy at gmail.com Mon Nov 10 18:51:18 2014 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Mon, 10 Nov 2014 17:51:18 +0200 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: References: Message-ID: <5460DEF6.6070204@gmail.com> You want to receive or send dtmf via mod_verto? On 10.11.14 17:04, Bal?zs S?ndor wrote: > Hi all! > > I would like to receive dtmf with mod_verto. > How can I do that? > > Best regards, > > > Bal?zs S?NDOR > > Software Developer > > > > > Virtual Call Center > > MUNICH | BUDAPEST > | WARSAW > > > Phone: +44 (0) 863 801 69 > > Web: www.virtual-call-center.eu > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/fe6669b6/attachment.html From rtreleaven at bunnykick.ca Mon Nov 10 19:10:29 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Mon, 10 Nov 2014 11:10:29 -0500 Subject: [Freeswitch-users] Freeswitch port numbers assigned to phones In-Reply-To: <5460B988.4040209@openlogic.com.au> References: <5460B988.4040209@openlogic.com.au> Message-ID: Dion, A endpoint registers with freeswitch to let freeswitch know how to reach it. The key piece of the registration record is the contact line Contact: "" The user is telling us his name and his ip address(street address) and suite(port number). The endpoint's port number is configured by the endpoint. Hope this helps. On Mon, Nov 10, 2014 at 8:11 AM, Dion Phillips wrote: > Hi All > > I am looking for some insight into FS port numbers. > > The default internal port number is 5060 and when the IP phones register > with FS, one of the phones gets assigned 5060 and the remaining phones get > a random port number assigned (I am using this command to see the port > numbers: "sofia status profile internal reg"). > > How is this random port number decided on? How can I tie a phone to a > specific port number? > > Thanks > Dion. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/01ce9e70/attachment.html From flokrrr at gmail.com Mon Nov 10 19:42:29 2014 From: flokrrr at gmail.com (Florent Krieg) Date: Mon, 10 Nov 2014 17:42:29 +0100 Subject: [Freeswitch-users] Freeswitch port numbers assigned to phones In-Reply-To: <5460B988.4040209@openlogic.com.au> References: <5460B988.4040209@openlogic.com.au> Message-ID: Hi Dion, The port you are seeing is the port used by the devices (phones/softphones/...) to send SIP messages to your server. I guess your phones are located behind a router on a local network. When the first phone sends SIP messages to your FS, it leaves the router with the port 5060 (it's up to the router to re-use 5060 or chose a random port from the beginning) but then for the other phones it has to map to other external ports, that's why a random one is used. Your router might be able to provide a way to configure NAT mappings manually, you should look that way. Florent 2014-11-10 14:11 GMT+01:00 Dion Phillips : > Hi All > > I am looking for some insight into FS port numbers. > > The default internal port number is 5060 and when the IP phones register > with FS, one of the phones gets assigned 5060 and the remaining phones get > a random port number assigned (I am using this command to see the port > numbers: "sofia status profile internal reg"). > > How is this random port number decided on? How can I tie a phone to a > specific port number? > > Thanks > Dion. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/6ec1722e/attachment.html From aqsyounas at gmail.com Mon Nov 10 19:43:41 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Mon, 10 Nov 2014 21:43:41 +0500 Subject: [Freeswitch-users] how to make ivr from json data Message-ID: I have ivr data in json format i want this data to be played in freeswitch like i used to do in asterisk with phpivr third party application. Does somebody know how to do so..or is there any application like phpivr in asterisk. https://code.google.com/p/phpivr/ Any help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/9a87e540/attachment.html From mbodbg at gmx.net Mon Nov 10 19:53:17 2014 From: mbodbg at gmx.net (mbo) Date: Mon, 10 Nov 2014 17:53:17 +0100 Subject: [Freeswitch-users] Freeswitch webRTC - audio DTLS key err Message-ID: <92D31E8B-D0B3-469C-B4EC-F0160C96B165@gmx.net> Hello, according to the instructions on https://freeswitch.org/confluence/display/FREESWITCH/WebRTC I have enabled webrtc on one Sip profile. I?ve added my own cerificate and installed in /usr/local/freeswitch/certs. I?m connecting to freeswitch using JsSip library. The signaling part works fine, I can see all SIP messages in the log, but then it fails to establish the audio/rtp connection: 2014-11-10 17:36:27.065688 [INFO] switch_core_media.c:5206 Skipping RTCP ICE (Same as RTP) 2014-11-10 17:36:27.065688 [INFO] switch_rtp.c:3065 Activate RTP/RTCP audio DTLS client 2014-11-10 17:36:27.065688 [ERR] switch_rtp.c:3117 audio DTLS key err [1] I?m a bit confused about this message, I thought all tls settings in a sip profile and the dtls-srtp.pem certificate is not relevant for webrtc / wss, or what does this error message want to tell me? Thanks Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/b71d0442/attachment-0001.html From steveayre at gmail.com Mon Nov 10 20:12:55 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 10 Nov 2014 17:12:55 +0000 Subject: [Freeswitch-users] C Dialplan processing routines In-Reply-To: References: Message-ID: Build it as a module. Every module under src/mod/ is an example Also see docs.freeswitch.org On 10 November 2014 15:18, Abdul Hakeem wrote: > Hello, > Could anyone give me an example on how to link a C code based routine to > invoke FS API's to process not just Dialplan or Chatplan but other events > such as Register, Reinvite etc ? > Thank you in advance. > Regards, > Abdul Hakeem > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/f4d16090/attachment.html From anthony.minessale at gmail.com Mon Nov 10 20:26:23 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 10 Nov 2014 11:26:23 -0600 Subject: [Freeswitch-users] Confusing about FreeSWITCH callID variables In-Reply-To: <54609d28.43c0c20a.1f0e.ffffe78a@mx.google.com> References: <545b7f4c.047eb40a.224c.ffff974a@mx.google.com> <545f8b31.a96db40a.58b0.3c2b@mx.google.com> <54609d28.43c0c20a.1f0e.ffffe78a@mx.google.com> Message-ID: Lesson 1 in FreeSWITCH would be never think of a call as one thing. Its always a union of 2 channels. SIP call ID is protocol specific to SIP and is distinct per leg of a call. FS is a b2bua so each call is 2 sip dialogs. Unique-ID is a FS specific ID also per call leg. Channel-Call-UUID is a field that is common between both legs of a bridged call. In the general case of an inbound leg bridging to an outbound leg, both channels will share the Channel Call ID and it will be the same as the Unique-ID of the inbound leg. You also have Channel-Bridge events which connect which 2 unique id's are currently bridged. On Mon, Nov 10, 2014 at 5:05 AM, Israel Fink wrote: > Regards my problem with the client, I already solved the problem in code. > To be clear about FS, is there any another variable that is unique for the > 'call' no matter which call leg? > > Israel Fink - Developer > Machshevet team > > *From:* Steven Ayre > *Sent:* ?Monday?, ?November? ?10?, ?2014 ?1?:?05? ?AM > *To:* freeswitch-users at lists.freeswitch.org > > sip_call_id is the Call-ID header of the SIP packet. It will usually be > different for the A-leg and B-leg of the call. However once in a call it > identifies the call and *must* stay the same. If you are seeing it change > then you're either looking at different legs of the call. > > On 9 November 2014 10:33, Israel Fink wrote: > >> Thanks for your clear answer. >> But I find thinks a bit different. we do collect the FS events during ESL >> mod event_socket, and find that the Unique-ID changes by channel. >> What I try to do is, we have a FS server, each call comes in via the >> server, there were made some logic like answer, greeting etc. and also >> saving the call in DB. from there it goes to a client extension to get the >> call, speak, hang up etc. when it comes to the client I need to know which >> call it is in the server and in the DB. >> The only variable the client gets from the server is sip_call_id, and now >> after your explanation I understand why, since this is outbound. >> My problem is that this variable also changes when bridging, it begins >> with one value then when bridging it changes to another value. the client >> gets the second value after bridging, and I cannot find this new value in >> early events like 'CHANNEL_CREATE' or 'CHANNEL_ORIGINATE', it only appears >> in later events like 'CHANNEL_PROGRESS' when the client already ringing, >> and this is a bit late. >> In short, I need a unique variable to can recognize the incoming call to >> the client with the incoming call to the server. >> >> Israel Fink - Developer >> Machshevet team >> >> *From:* Brian West >> *Sent:* ?Friday?, ?November? ?7?, ?2014 ?12?:?23? ?AM >> *To:* freeswitch-users at lists.freeswitch.org >> >> 1. Unique-ID is the session UUID, this won't change at all during the >> session (aka call). >> >> 2. Channel-Call-UUID is either the same as Unique-ID or the Unique-ID of >> the session you're bridged to by looking at the source code. >> >> 3. sip_call_id will be the sip call_id of an inbound/outbound SIP call, >> More than likely the outbound would match Unique-ID and on inbound it would >> be what ever the remote party sets/sends. >> >> The bigger question is what exactly are you trying to solve/understand? >> >> On Thu, Nov 6, 2014 at 7:52 AM, Israel Fink wrote: >> >>> I'm a bit confusing about the channel variables that represent the >>> call ID. >>> >>> I find that there are three variables, 1) Unique-ID 2) Channel-Call-UUID >>> 3) variable_sip_call_id, what is the different between this. >>> >>> The value of Channel-Call-UUID seems to be stable during the entire >>> call, the value of Unique-ID in the begin it is the same as >>> Channel-Call-UUID but when bridging it changes it value to another ID, then >>> it comes back to the beginning value, i.m not clear when and why. >>> >>> The variable variable_sip_call_id many time don't have any value, and >>> also it changes the value when bridging and then comes back to the previous >>> value. >>> >>> I have looked for an explanation in FreeSWITCH wiki, but don't find. >>> >>> In the mailing list a have found an explanation here: >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2013-August/099035.html, >>> but still not enough. >>> Can someone give a good explanation about this variables,what is their >>> purpose, and so on. >>> >>> >>> Israel Fink - Developer >>> Machshevet team >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/f7666a5f/attachment-0001.html From anthony.minessale at gmail.com Mon Nov 10 20:32:55 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 10 Nov 2014 11:32:55 -0600 Subject: [Freeswitch-users] Freeswitch webRTC - audio DTLS key err In-Reply-To: <92D31E8B-D0B3-469C-B4EC-F0160C96B165@gmx.net> References: <92D31E8B-D0B3-469C-B4EC-F0160C96B165@gmx.net> Message-ID: You could run tshark on a terminal on the box and filter for dtls traffic to get a better idea. Contrary to your statement, the dtls-srtp.pem is only relevant to WebRTC. You can try deleting that file and let FS generate a new one by restarting it. Also check the date and time on your box to make sure its correct. On Mon, Nov 10, 2014 at 10:53 AM, mbo wrote: > Hello, > > according to the instructions on > https://freeswitch.org/confluence/display/FREESWITCH/WebRTC I have > enabled webrtc on one Sip profile. I?ve added my own cerificate and > installed in /usr/local/freeswitch/certs. I?m connecting to freeswitch > using JsSip library. The signaling part works fine, I can see all SIP > messages in the log, but then it fails to establish the audio/rtp > connection: > > 2014-11-10 17:36:27.065688 [INFO] switch_core_media.c:5206 Skipping RTCP > ICE (Same as RTP) > 2014-11-10 17:36:27.065688 [INFO] switch_rtp.c:3065 Activate RTP/RTCP > audio DTLS client > 2014-11-10 17:36:27.065688 [ERR] switch_rtp.c:3117 audio DTLS key err [1] > > > I?m a bit confused about this message, I thought all tls settings in a sip > profile and the dtls-srtp.pem certificate is not relevant for webrtc / wss, > or what does this error message want to tell me? > > Thanks > > Markus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/9163f6b8/attachment.html From aviv at sent.com Mon Nov 10 22:17:06 2014 From: aviv at sent.com (Aviv Shaham) Date: Mon, 10 Nov 2014 12:17:06 -0700 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415475715878.91825@nexxuspg.com> References: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> <1415475715878.91825@nexxuspg.com> Message-ID: <1415647026.1146908.189312005.4EEC8C03@webmail.messagingengine.com> Hi Manish, First, no need to set nexmo_forwarded_for for outbound here, if you re-read my previous email you'll see that it was meant to be used for Nexmo DIDs you use to direct origination traffic into FS. As for the error you are getting with this dialplan, you need to remove "@sip.nexmo.com:5080" from your origination string. Hope it helps, Aviv On Fri, Nov 7, 2014, at 11:10 PM, Manish Talwar wrote: > Hi, > > Thanks for your suggestion, I have make these changes and > removed the L16 codec from request now. I have set "absolute_codec_string" and "nexmo_forwarded_for" and its not throwing any error message in SIP trace now. > > But still, I am not able to make a call on my mobile number > "*1919818753995*". Its show message on FreeSwitch log as > "[RECOVERY_ON_TIMER_EXPIRE]" and hangup the freeswitch call. Also, > there is no log created on Nexmo dashboard for this call's. > > I am sending my call request to Nexmo from FreeSwitch by dialplan as. > > ** * field="destination_number" expression="^19(1\d{10})$">* * application="set" data="absolute_codec_string=PCMU,GSM"/>* * application="set" data="nexmo_forwarded_for=$1"/>* * application="bridge" > data="{origination_caller_id_name='18188535351',ignore_early_media=true}sofia/gateway/nexmo/$1 at sip.nexmo.com:5080"/> * > * * * * > > > Please find the attached SipTrace file now and let me know what I need > to update now. > > In this log, values passed in "From" and "To" attribute as: ** > ** > *From: "18188535351" ;tag=D8g4a5NvH4emF* * > To: * > > I feel there might be some wrong data passed in "To" attribute and it > might expecting mobile number "19818753995" only instead on SIP value. > Please suggest about these setting also. > > Thanks, > > Regards, > Manish Talwar > > > *From:* Aviv Shaham *Sent:* 07 November 2014 21:48 > *To:* freeswitch-users at lists.freeswitch.org *Subject:* Re: > [Freeswitch-users] SIP trunking with Nexmo > > Hi Manish, > > Nexmo doesn't seem to handle it well if your first specified > codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. > > Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. > > Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: > nexmo_12121115555 at your-server.com and then have a dialplan such as: > > expression="^nexmo_(\d+)$"> application="set" data="nexmo_forwarded_for=$1"/> application="lua" data="nexmo_handler.lua"/> > > The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. > > Let me know if you are having any other issues. > > Aviv > > > On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote: >> Hi, >> >> Thanks for your suggestion, I have tried it and I am able to do a >> Inbound call via Nexmo now. But still I am not able to make any >> outbound call from my application. >> >> I have checked the FreeSwitch log by siptrace enable and found that >> my call was terminated with a SIP message as " >> *IP/2.0 407 Proxy Authentication Required*". >> >> Please find the siptrace log for my call as an attachment. and let me >> know what changes or configuration I need to make for Proxy >> Authentication Header. >> >> Thanks, >> >> Regards, >> Manish Talwar >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org >> on behalf of Aviv Shaham >> *Sent:* 06 November 2014 14:39 *To:* >> freeswitch-users at lists.freeswitch.org *Subject:* Re: >> [Freeswitch-users] SIP trunking with Nexmo >> >> Hi Manish, >> >> Nexmo expects your API KEY to be in the From header. To set the >> caller ID you will need to use "caller-id-name". Good timing btw, I >> just posted a reply to a similar question on Quora. Have a look: >> http://qr.ae/DEbk2 - also covers Plivo. >> >> Aviv >> >> >> On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: >>> Hi, >>> >>> I have make a SIP Trunking (gateway) in FreeSwitch for connecting >>> Nexmo via bridge. I have added this Nexmo file under >>> "*\FreeSWITCH\conf\sip_profiles\external*" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: >>> >>> >>> Name Type Data State >>> ================================================================================================ >>> external-ipv6 profile >>> sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING >>> (0) 192.168.1.140 alias internal ALIASED external profile >>> sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com >>> gateway sip:joeuser at example.com NOREG external::sip.nexmo.com >>> gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED >>> internal-ipv6 profile >>> sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING >>> (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) >>> ================================================================================================ >>> 4 profiles 1 alias >>> >>> But when I send the request to FreeSwitch by Dial command as: >>> *>> type="xml/freeswitch-httapi">>> application="set" data="sip_h_api_key=b9c280dd" />>> application="set" data="sip_h_api_secret=7678b8c4" />>> application="set" data="sip_h_to=919818753995" />>> *caller-id-number="18188535351" context="default" Dialplan="XML" >>> >919818753995* >>> >>> >>> here, *18188535351* = Nexmo virtual number for connecting call. >>> *919818753995* = mobile number where I am looking for making a call. >>> >>> It will not connected to Nexmo and call will be terminated with >>> message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 >>> Originate Failed. Cause: NORMAL_UNSPECIFIED >>> >>> Please find the FreeSwitch call Log and Nexmo Gateway (which I have >>> added in freeswitch conf external folder) as an attachment. >>> >>> Please let me know whether I am doing SIP trunking in correct way or >>> need to change something. >>> >>> Also, Please suggest me what will be my next step for making a call >>> on mobile by this ways. >>> >>> Thanks, >>> >>> Regards, Manish Talwar >>> >>> ___________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites http://www.freeswitch.org >>> http://confluence.freeswitch.org http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org Email had 2 attachments: >>> * FsCall.txt >>> 15k (text/plain) >>> * Nexmo.xml >>> 3k (text/xml) >> >> ___________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites http://www.freeswitch.org >> http://confluence.freeswitch.org http://www.cluecon.com >> >> FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org Email had 1 attachment: >> * SipTrace.txt >> 9k (text/plain) > > ___________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites http://www.freeswitch.org > http://confluence.freeswitch.org http://www.cluecon.com > > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Email had 1 attachment: > * SipTrace.txt 16k (text/plain) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/d05513c7/attachment-0001.html From aqsyounas at gmail.com Mon Nov 10 23:20:26 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 11 Nov 2014 01:20:26 +0500 Subject: [Freeswitch-users] phpivr/SynIVR like application in freeswitch Message-ID: I have ivr data into json format. I want to execute that data in freeswitch.Is there any application available to do so.? Or how can i do so? Like we have phpivr/SynIVR in asterisk. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/5d0b540a/attachment.html From mike at jerris.com Mon Nov 10 23:27:03 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Nov 2014 15:27:03 -0500 Subject: [Freeswitch-users] phpivr/SynIVR like application in freeswitch In-Reply-To: References: Message-ID: <81CAE088-618A-4348-8A08-24E962693CCF@jerris.com> mod_httapi is similar. > On Nov 10, 2014, at 3:20 PM, Aqs Younas wrote: > > I have ivr data into json format. I want to execute that data in freeswitch.Is there any application available to do so.? > > Or how can i do so? > > Like we have phpivr/SynIVR in asterisk. From aqsyounas at gmail.com Mon Nov 10 23:32:24 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 11 Nov 2014 01:32:24 +0500 Subject: [Freeswitch-users] phpivr/SynIVR like application in freeswitch In-Reply-To: <81CAE088-618A-4348-8A08-24E962693CCF@jerris.com> References: <81CAE088-618A-4348-8A08-24E962693CCF@jerris.com> Message-ID: Thanks for your reply. After having a look at mod_httapi i get back to you. On 11 November 2014 01:27, Michael Jerris wrote: > mod_httapi is similar. > > > On Nov 10, 2014, at 3:20 PM, Aqs Younas wrote: > > > > I have ivr data into json format. I want to execute that data in > freeswitch.Is there any application available to do so.? > > > > Or how can i do so? > > > > Like we have phpivr/SynIVR in asterisk. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/6321c92d/attachment.html From balazs.sandor at virtual-call-center.hu Tue Nov 11 00:42:59 2014 From: balazs.sandor at virtual-call-center.hu (=?UTF-8?B?U8OhbmRvciBCYWzDoXpz?=) Date: Mon, 10 Nov 2014 22:42:59 +0100 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: <5460DEF6.6070204@gmail.com> References: <5460DEF6.6070204@gmail.com> Message-ID: Receive On 10 Nov 2014 17:17, "Igor Olhovskiy" wrote: > You want to receive or send dtmf via mod_verto? > > On 10.11.14 17:04, Bal?zs S?ndor wrote: > > Hi all! > > I would like to receive dtmf with mod_verto. > How can I do that? > > Best regards, > > > Bal?zs S?NDOR > > Software Developer > > > > > Virtual Call Center > > MUNICH | BUDAPEST > | WARSAW > > > Phone: +44 (0) 863 801 69 > > Web: www.virtual-call-center.eu > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/a6bcb9dd/attachment.html From mike at jerris.com Tue Nov 11 00:48:26 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Nov 2014 16:48:26 -0500 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: References: <5460DEF6.6070204@gmail.com> Message-ID: <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> and by that you mean you want to receive on the server, send from the client? That is built in to mod_verto already, and is functional in the demo at webrtc.freeswitch.org > On Nov 10, 2014, at 4:42 PM, S?ndor Bal?zs wrote: > > Receive > > On 10 Nov 2014 17:17, "Igor Olhovskiy" > wrote: > You want to receive or send dtmf via mod_verto? > > On 10.11.14 17:04, Bal?zs S?ndor wrote: >> Hi all! >> >> I would like to receive dtmf with mod_verto. >> How can I do that? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/f0d93b85/attachment.html From karl-theo_hofer at inteli-sim.com Tue Nov 11 01:11:32 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Mon, 10 Nov 2014 23:11:32 +0100 Subject: [Freeswitch-users] no ring back tone Message-ID: <54613814.3060504@inteli-sim.com> Hi there please be gentle with me but i can not find any help in the wiki to solve my problem this is a desperat try to get some answers to the following problem freeswitch is a configured as call back server FS originates the A-leg when answered the B-leg is originated. after A-leg has answered I like to play a short welcome message After the welcome mesage is played the international, original ringbacktone (telco carrier RBT) from the B-leg shall be passed through and the A-leg user should be able to hear it. This is important to hear some carrier specific announcements or coloured/personalized ring back. Right now we play our own ringbacktone even though the carrier provides us with a 183 sdp I did play with the usual Channel variables but as far as I understood the passing through of the ringbacktone is default behaviour. -- With best regards kT From aademattia at comcast.net Tue Nov 11 01:24:42 2014 From: aademattia at comcast.net (=?utf-8?B?YWFkZW1hdHRpYUBjb21jYXN0Lm5ldA==?=) Date: Mon, 10 Nov 2014 17:24:42 -0500 Subject: [Freeswitch-users] =?utf-8?b?aGFzaF9yZW1vdGUgPGxpc3Q+fDxraWxsPiBb?= =?utf-8?b?bmFtZV18PHJlc2Nhbj4=?= Message-ID: I could use this information too. Sent from my HTC ----- Reply message ----- From: "Andre Demattia" To: "FreeSWITCH Users Help" Subject: [Freeswitch-users] hash_remote | [name]| Date: Sun, Nov 9, 2014 7:39 PM Hi, What would the syntax look like to connect more than one FreeSwitch Limit hash_remote | [name]| Thanks Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141110/42cbef1a/attachment.html From ssinyagin at gmail.com Tue Nov 11 02:31:54 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 11 Nov 2014 00:31:54 +0100 Subject: [Freeswitch-users] no ring back tone In-Reply-To: <54613814.3060504@inteli-sim.com> References: <54613814.3060504@inteli-sim.com> Message-ID: if you don't set ignore_early_media to true, "bridge" should deliver you the B-leg's early media before it's answered. For example, if I call a mobile number and the calle is unreachable, I hear the operator's greeting, but the call is not answered. Also makes sense to check if your ITSP is sending you the early media correctly. Wireshark is your best friend. On Mon, Nov 10, 2014 at 11:11 PM, kthofer wrote: > Hi there > > please be gentle with me but i can not find any help in the wiki to > solve my problem > this is a desperat try to get some answers to the following problem > freeswitch is a configured as call back server > FS originates the A-leg when answered the B-leg is originated. > after A-leg has answered I like to play a short welcome message > After the welcome mesage is played the international, original > ringbacktone (telco carrier RBT) from the B-leg shall be passed through > and the A-leg user should be able to hear it. > This is important to hear some carrier specific announcements or > coloured/personalized ring back. > Right now we play our own ringbacktone even though the carrier provides > us with a 183 sdp > I did play with the usual Channel variables > but as far as I understood the passing through of the ringbacktone is > default behaviour. > > > > > > > > > -- > With best regards > kT > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ssinyagin at gmail.com Tue Nov 11 02:38:27 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 11 Nov 2014 00:38:27 +0100 Subject: [Freeswitch-users] phpivr/SynIVR like application in freeswitch In-Reply-To: References: Message-ID: there's plenty of different ways to implement an IVR with FreeSWITCH, but as far as I know, this particular JSON format is not yet implemented by anyone. Probably there was never a need. Why don't you simply re-implement your IVR in one of the FreeSWITCH ways? This is to start with: https://wiki.freeswitch.org/wiki/IVR_Menu You can also implement it in Lua or Perl or Python, or (better) create an ESL application and control the IVR from there. You can even implement the phpivr language if you really need to. On Mon, Nov 10, 2014 at 9:20 PM, Aqs Younas wrote: > I have ivr data into json format. I want to execute that data in > freeswitch.Is there any application available to do so.? > > Or how can i do so? > > Like we have phpivr/SynIVR in asterisk. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lakindia89 at gmail.com Tue Nov 11 07:20:42 2014 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 11 Nov 2014 09:50:42 +0530 Subject: [Freeswitch-users] no ring back tone In-Reply-To: <54613814.3060504@inteli-sim.com> References: <54613814.3060504@inteli-sim.com> Message-ID: I think you need to set ignore_early_media=false and bridge_early_media=true, which will pass the ringback tone from B leg to A leg. On Tue, Nov 11, 2014 at 3:41 AM, kthofer wrote: > Hi there > > please be gentle with me but i can not find any help in the wiki to > solve my problem > this is a desperat try to get some answers to the following problem > freeswitch is a configured as call back server > FS originates the A-leg when answered the B-leg is originated. > after A-leg has answered I like to play a short welcome message > After the welcome mesage is played the international, original > ringbacktone (telco carrier RBT) from the B-leg shall be passed through > and the A-leg user should be able to hear it. > This is important to hear some carrier specific announcements or > coloured/personalized ring back. > Right now we play our own ringbacktone even though the carrier provides > us with a 183 sdp > I did play with the usual Channel variables > but as far as I understood the passing through of the ringbacktone is > default behaviour. > > > > > > > > > -- > With best regards > kT > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/aabdc9a7/attachment.html From dujinfang at gmail.com Tue Nov 11 09:09:06 2014 From: dujinfang at gmail.com (Seven Du) Date: Tue, 11 Nov 2014 14:09:06 +0800 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: Message-ID: <107291DC4A8A424280FDB589D568639B@gmail.com> No. Maybe bounty it? On Monday, November 10, 2014 at 8:53 PM, ik wrote: > Hello, > > I could not find any documentation on this subject. > I require to have a recording of an a-law wav file (RIFF (little-endian) data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). > > Can I set a variable that tells the record command what type of codec to store the file with ? > > I know how to convert a normal wav recording into it using ffmpeg (for example): > > $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav > > But I wish to avoid it, if I can use the record app, or use the uuid_record API for it. > > Thanks, > Ido > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/55cdc5ed/attachment.html From dujinfang at gmail.com Tue Nov 11 09:22:37 2014 From: dujinfang at gmail.com (Seven Du) Date: Tue, 11 Nov 2014 14:22:37 +0800 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: References: Message-ID: <882AF05B1A21418D87AAF24C2F0CC49E@gmail.com> not sure how to do that in perl, but you should subscribe to the CUSTOM conference::maintenance event or ALL event and you could get that with $e->getHeader("Member-ID") -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, November 7, 2014 at 10:02 PM, ???? wrote: > Thank you Mr.Seven Du > > I got the sound !. it was my mistake. > because of localhost. It should be IP. > $con = new ESL::ESLconnection("localhost", 8021, ClueCon); > > However, I still have problem. > > Let me know how to get 'member_id' below. in perl. > i tried below. > $e->getHeader("Caller-Caller-ID-Name"); > $e->getHeader("Caller-Caller-ID-Number"); > > > conference play [|] > > thank you. > > > On Wed, Nov 5, 2014 at 10:52 AM, Seven Du wrote: > > I don?t quite understand you but you should not put SPACE between % and s. Try to paste the real code and debug logs on pastebin to get better help. > > > > > > On Tuesday, November 4, 2014 at 11:51 PM, ???? wrote: > > > > > Dear Stanislav Sinyagin > > > > > > Thank you for your help. > > > > > > ????????????????????????????? > > > > > > ???Perl??? > > > > > > ?????????????????????????wav?????????????????? > > > > > > ????????????????????? > > > my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > > > > ???$ con-> bgapi ($ api_cmd); ????????????? > > > > > > fs_cli ? ?conference ************* play? ????????????????????????? > > > > > > ????????????? > > > > > > ??????????? > > > > > > > > > > > > On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin wrote: > > > > Dear Matsumoto-san, > > > > > > > > I think it will be easier if you write in Japanese, then it will be clear how we could help. I know a few Japanese-speaking colleagues who may help in communicating. > > > > > > > > > > > > > > > > > > > > On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > > > > > Hello > > > > > > > > > > I have two issues. > > > > > > > > > > I am writing in Perl. > > > > > > > > > > While 2 people are talking in a conference room, the one person want to play the sound. > > > > > > > > > > In "Caller-Username", can you get useless. > > > > > I have tried the above but, Member: it will not become a *** not found.. > > > > > > > > > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > > > > > > > > > > > > > I want to play the above in the background > > > > > It can not play in the next program. > > > > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader ("Conference-Name"), /etc/a.wav); > > > > > > > > > > $ con-> bgapi ($ api_cmd); > > > > > > > > > > Best regards > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://confluence.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://confluence.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://confluence.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/97eed619/attachment-0001.html From balazs.sandor at virtual-call-center.hu Tue Nov 11 11:17:43 2014 From: balazs.sandor at virtual-call-center.hu (=?UTF-8?B?U8OhbmRvciBCYWzDoXpz?=) Date: Tue, 11 Nov 2014 09:17:43 +0100 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> References: <5460DEF6.6070204@gmail.com> <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> Message-ID: I call a mobile phone from browser with mod_verto, and I would like to receive dtmfs in the browser (sent from mobile phone) Sorry if I was not clear enough. ?dv?zlettel: S?ndor Bal?zs Szoftver fejleszt? Virtual Call Center MUNICH | BUDAPEST | WARSAW Telefon: +36 1 999 7400 Web: www.virtual-call-center.hu 2014-11-10 22:48 GMT+01:00 Michael Jerris : > and by that you mean you want to receive on the server, send from the > client? That is built in to mod_verto already, and is functional in the > demo at webrtc.freeswitch.org > > On Nov 10, 2014, at 4:42 PM, S?ndor Bal?zs < > balazs.sandor at virtual-call-center.hu> wrote: > > Receive > On 10 Nov 2014 17:17, "Igor Olhovskiy" wrote: > >> You want to receive or send dtmf via mod_verto? >> >> On 10.11.14 17:04, Bal?zs S?ndor wrote: >> >> Hi all! >> >> I would like to receive dtmf with mod_verto. >> How can I do that? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/fcd05df9/attachment.html From idokan at gmail.com Tue Nov 11 12:52:41 2014 From: idokan at gmail.com (ik) Date: Tue, 11 Nov 2014 11:52:41 +0200 Subject: [Freeswitch-users] setting recording codec In-Reply-To: <107291DC4A8A424280FDB589D568639B@gmail.com> References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: Thinking in learning better the source code, and might create my first patch to FS :) On Tue, Nov 11, 2014 at 8:09 AM, Seven Du wrote: > No. Maybe bounty it? > > On Monday, November 10, 2014 at 8:53 PM, ik wrote: > > Hello, > > I could not find any documentation on this subject. > I require to have a recording of an a-law wav file (RIFF (little-endian) > data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). > > Can I set a variable that tells the record command what type of codec to > store the file with ? > > I know how to convert a normal wav recording into it using ffmpeg (for > example): > > $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav > > But I wish to avoid it, if I can use the record app, or use the > uuid_record API for it. > > Thanks, > Ido > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/d7e83f08/attachment.html From ssinyagin at gmail.com Tue Nov 11 13:21:53 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Tue, 11 Nov 2014 11:21:53 +0100 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: as far as I understand, FreeSWITCH chooses the WAV parameters which are matching the current channel sampling frequency and codec. For example, if you record a G722 call, you would get a 16kHZ WAV file. Forcing it into one specific encoding would add real-time CPU load. I think it's still more preferable to run a post-processing job in low priority -- this way you ensure that the ongoing calls get the best serving. You can easily catch the event of call ending and trigger the conversion job, by listening to the events via ESL connection. On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: > Thinking in learning better the source code, and might create my first patch > to FS :) > > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du wrote: >> >> No. Maybe bounty it? >> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >> >> Hello, >> >> I could not find any documentation on this subject. >> I require to have a recording of an a-law wav file (RIFF (little-endian) >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >> >> Can I set a variable that tells the record command what type of codec to >> store the file with ? >> >> I know how to convert a normal wav recording into it using ffmpeg (for >> example): >> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >> >> But I wish to avoid it, if I can use the record app, or use the >> uuid_record API for it. >> >> Thanks, >> Ido >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From idokan at gmail.com Tue Nov 11 13:33:50 2014 From: idokan at gmail.com (ik) Date: Tue, 11 Nov 2014 12:33:50 +0200 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: In this specific case, I'm using g711, but can have either a-law or u-law but the service that requires the recording understand only a-law, at the moment I'm using ffmpeg to convert after the call ended, and then send it to that service. Thanks Ido On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" wrote: > as far as I understand, FreeSWITCH chooses the WAV parameters which > are matching the current channel sampling frequency and codec. For > example, if you record a G722 call, you would get a 16kHZ WAV file. > > Forcing it into one specific encoding would add real-time CPU load. I > think it's still more preferable to run a post-processing job in low > priority -- this way you ensure that the ongoing calls get the best > serving. > > You can easily catch the event of call ending and trigger the > conversion job, by listening to the events via ESL connection. > > > > > On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: > > Thinking in learning better the source code, and might create my first > patch > > to FS :) > > > > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du wrote: > >> > >> No. Maybe bounty it? > >> > >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: > >> > >> Hello, > >> > >> I could not find any documentation on this subject. > >> I require to have a recording of an a-law wav file (RIFF (little-endian) > >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). > >> > >> Can I set a variable that tells the record command what type of codec to > >> store the file with ? > >> > >> I know how to convert a normal wav recording into it using ffmpeg (for > >> example): > >> > >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav > >> > >> But I wish to avoid it, if I can use the record app, or use the > >> uuid_record API for it. > >> > >> Thanks, > >> Ido > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/6488aad1/attachment-0001.html From dion at openlogic.com.au Tue Nov 11 11:25:13 2014 From: dion at openlogic.com.au (Dion Phillips) Date: Tue, 11 Nov 2014 16:25:13 +0800 Subject: [Freeswitch-users] UDP vs UDP-NAT Message-ID: <5461C7E9.5090209@openlogic.com.au> Hi All I have a FS server in the cloud and have phones registering with FS from 2 different locations. From one location, the registration is UDP-NAT and just UDP from the second location. In the FS log, calls to the UDP connected phones work and they come up with an external IP address (sofia_glue.c:1232 sofia/internal/sip:1002 at 203.59.xxx.xxx:5060 sending invite) while the phones on the UDP-NAT site don't work because FS is trying to connect to them via the internal IP address (sofia_glue.c:1232 sofia/internal/sip:1000 at 192.168.0.111:5060 sending invite). The phones on both sites are behind a very standard modem with no port forwarding or NAT setting. What is wrong and how would I fix it? Thanks in advance. Dion. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/7e378613/attachment-0001.html From manish.talwar at nexxuspg.com Tue Nov 11 12:13:06 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Tue, 11 Nov 2014 09:13:06 +0000 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415647026.1146908.189312005.4EEC8C03@webmail.messagingengine.com> References: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> <1415475715878.91825@nexxuspg.com>, <1415647026.1146908.189312005.4EEC8C03@webmail.messagingengine.com> Message-ID: <1415745838465.55443@nexxuspg.com> Hi, Thanks a lot, I am able to make a outbound call by these settings. I have set "absolute_codec_string" as "PCMA,PCMU" and remove "nexmo_forwarded_for" from my dialplan. I have tried this outbound call by "FsClient" and "Wizton" application. Call is running fine with "FsClient" but unable to received the call by "Wizton". In Wizton, Its tried to call on my mobile with similar logs in "FS_CLI" as calling from "FsClient" but after few seconds call was hangup with message as "Originate Failed. Cause: ORIGINATOR_CANCEL". I feel it might be some "codec" configuration problem only, Please find the "FsClient" and "Wizton" log file as an attachment. With, "FsClient" log file I am able to receive the call on my mobile (+919818753995). Also, when its calling Outbound call by "FsClient" then there was no ring sound came on "FsClient" but call was coming on mobile. Please suggest me, what I need to do for Ring sound while its calling a mobile and also Is there any other setting required while calling from "Wizton" or any other medium (like mobile, phone etc). Thanks a lot. Regards, Manish Talwar Also, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 11 November 2014 00:47 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, First, no need to set nexmo_forwarded_for for outbound here, if you re-read my previous email you'll see that it was meant to be used for Nexmo DIDs you use to direct origination traffic into FS. As for the error you are getting with this dialplan, you need to remove "@sip.nexmo.com:5080" from your origination string. Hope it helps, Aviv On Fri, Nov 7, 2014, at 11:10 PM, Manish Talwar wrote: Hi, Thanks for your suggestion, I have make these changes and removed the L16 codec from request now. I have set "absolute_codec_string" and "nexmo_forwarded_for" and its not throwing any error message in SIP trace now. But still, I am not able to make a call on my mobile number "1919818753995". Its show message on FreeSwitch log as "[RECOVERY_ON_TIMER_EXPIRE]" and hangup the freeswitch call. Also, there is no log created on Nexmo dashboard for this call's. I am sending my call request to Nexmo from FreeSwitch by dialplan as. Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" ;tag=D8g4a5NvH4emF To: I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham Sent: 07 November 2014 21:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555 at your-server.com and then have a dialplan such as: The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. Let me know if you are having any other issues. Aviv On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote: Hi, Thanks for your suggestion, I have tried it and I am able to do a Inbound call via Nexmo now. But still I am not able to make any outbound call from my application. I have checked the FreeSwitch log by siptrace enable and found that my call was terminated with a SIP message as " IP/2.0 407 Proxy Authentication Required". Please find the siptrace log for my call as an attachment. and let me know what changes or configuration I need to make for Proxy Authentication Header. Thanks, Regards, Manish Talwar ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 06 November 2014 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo. Aviv On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: Hi, I have make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: 919818753995 here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 16k (text/plain) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/e9d71855/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FsClient_Log.docx Type: application/vnd.openxmlformats-officedocument.wordprocessingml.document Size: 17265 bytes Desc: FsClient_Log.docx Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/e9d71855/attachment-0002.bin -------------- next part -------------- A non-text attachment was scrubbed... Name: Wizton_Log.docx Type: application/vnd.openxmlformats-officedocument.wordprocessingml.document Size: 16899 bytes Desc: Wizton_Log.docx Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/e9d71855/attachment-0003.bin From eduardo.alonso at quobis.com Tue Nov 11 14:33:27 2014 From: eduardo.alonso at quobis.com (Eduardo Alonso) Date: Tue, 11 Nov 2014 12:33:27 +0100 Subject: [Freeswitch-users] Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit Message-ID: Dear list: This is my first mail to the list. I hope that my data and email are well formed. We are using freeswitch as voicemail in a production environment and every week we obtain a core dump like the attached file. The error occurs in the CoreSession::recordFile function when the customer try to record a greeting message. As is normal it's not possible to attach the entire core file, as well I attached the backtrace following the debugging guide in the wiki: https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. I hope that the attached file help you to find the error. If more information or testing is needed please let me know. I'm looking for a similar core in the list, but all errors related with this issue are different, from my point of view. Thank you in advance for your support. Cheers and best regards. -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis | e: eduardo.alonso at quobis.com | t: +34902999465 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/792bea7e/attachment-0001.html -------------- next part -------------- GNU gdb (GDB) 7.4.1-debian Copyright (C) 2012 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". For bug reporting instructions, please see: ... Reading symbols from /usr/local/freeswitch/bin/freeswitch...done. [New LWP 12924] [New LWP 10495] [New LWP 10502] [New LWP 10540] [New LWP 10788] [New LWP 10503] [New LWP 10507] [New LWP 10786] [New LWP 10498] [New LWP 13119] [New LWP 10493] [New LWP 10020] [New LWP 25300] [New LWP 10491] [New LWP 25302] [New LWP 10789] [New LWP 10781] [New LWP 13115] [New LWP 10541] [New LWP 10791] [New LWP 10778] [New LWP 10790] [New LWP 13345] [New LWP 10787] [New LWP 10784] [New LWP 25303] [New LWP 10542] [New LWP 10506] [New LWP 10494] [New LWP 10782] [New LWP 10785] [New LWP 25301] [New LWP 10783] [New LWP 10504] [New LWP 13344] [New LWP 13346] [New LWP 13327] warning: Can't read pathname for load map: Error de entrada/salida. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/freeswitch/bin/freeswitch -u freeswitch -g freeswitch -rp -nc -nonat'. Program terminated with signal 11, Segmentation fault. #0 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 1151 end_allow_threads(); ------------------------------------------------------------------------------ (gdb) set pagination off ------------------------------------------------------------------------------ (gdb) info threads Id Target Id Frame 37 Thread 0x7f43d0874700 (LWP 13327) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 36 Thread 0x7f43d08b0700 (LWP 13346) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 35 Thread 0x7f43d0928700 (LWP 13344) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 34 Thread 0x7f43e09e7700 (LWP 10504) 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 33 Thread 0x7f43d85d3700 (LWP 10783) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 32 Thread 0x7f43e1cfc700 (LWP 25301) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 31 Thread 0x7f43d8157700 (LWP 10785) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 30 Thread 0x7f43da710700 (LWP 10782) 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 29 Thread 0x7f43e1e42700 (LWP 10494) 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 28 Thread 0x7f43e09ab700 (LWP 10506) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 27 Thread 0x7f43db7eb700 (LWP 10542) 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 26 Thread 0x7f43d0a54700 (LWP 25303) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 25 Thread 0x7f43d8193700 (LWP 10784) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 24 Thread 0x7f43d80df700 (LWP 10787) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 23 Thread 0x7f43d09dc700 (LWP 13345) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 22 Thread 0x7f43d0b84700 (LWP 10790) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 21 Thread 0x7f43e0047700 (LWP 10778) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 20 Thread 0x7f43d0b48700 (LWP 10791) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 19 Thread 0x7f43db827700 (LWP 10541) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 18 Thread 0x7f43d07d9700 (LWP 13115) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 17 Thread 0x7f43da6d4700 (LWP 10781) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 16 Thread 0x7f43d8067700 (LWP 10789) 0x00007f43e3e0b3cd in accept () from /lib/x86_64-linux-gnu/libpthread.so.0 15 Thread 0x7f43d0a18700 (LWP 25302) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 14 Thread 0x7f43e53b6760 (LWP 10491) 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 13 Thread 0x7f43d0ad0700 (LWP 25300) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 12 Thread 0x7f43d0b0c700 (LWP 10020) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 11 Thread 0x7f43e52ee700 (LWP 10493) 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 10 Thread 0x7f43d09a0700 (LWP 13119) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 9 Thread 0x7f43e1cc0700 (LWP 10498) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 8 Thread 0x7f43d811b700 (LWP 10786) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 7 Thread 0x7f43e00c1700 (LWP 10507) 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 6 Thread 0x7f43e0c0e700 (LWP 10503) 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 5 Thread 0x7f43d80a3700 (LWP 10788) 0x00007f43e3e0b18d in read () from /lib/x86_64-linux-gnu/libpthread.so.0 4 Thread 0x7f43e0084700 (LWP 10540) 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 3 Thread 0x7f43e10d5700 (LWP 10502) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 2 Thread 0x7f43e1da3700 (LWP 10495) 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 * 1 Thread 0x7f43d0a91700 (LWP 12924) 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 ------------------------------------------------------------------------------ (gdb) bt #0 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 #1 0x00007f43d243b826 in _wrap_CoreSession_recordFile (L=0x7f43cc25d370) at mod_lua_wrap.cpp:4947 #2 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43cc25d370, func=0x7f43cce39e80, nresults=nresults at entry=0) at ldo.c:319 #3 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43cc25d370, nexeccalls=nexeccalls at entry=1) at lvm.c:587 #4 0x00007f43d244538d in luaD_call (L=0x7f43cc25d370, func=0x7f43ccd1a6d0, nResults=) at ldo.c:377 #5 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43cc25d370, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0a90370) at ldo.c:116 #6 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43cc25d370, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0a90370, old_top=, ef=) at ldo.c:463 #7 0x00007f43d2442be1 in lua_pcall (L=0x7f43cc25d370, nargs=0, nresults=0, errfunc=) at lapi.c:821 #8 0x00007f43d24277d6 in docall (L=0x7f43cc25d370, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 #9 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43cc25d370, input_code=, input_code at entry=0x7f43dc513c10 "vm_check_msgs.lua") at mod_lua.cpp:195 #10 0x00007f43d24288ed in lua_function (session=0x7f43ccd3c5b8, data=) at mod_lua.cpp:476 #11 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43ccd3c5b8, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584") at src/switch_core_session.c:2809 #12 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43ccd3c5b8, app=0x7f43d631e410 "lua", arg=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", flags=flags at entry=0x0) at src/switch_core_session.c:2684 #13 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:230 #14 switch_core_session_run (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:481 #15 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43ccd3c5b8) at src/switch_core_session.c:1542 #16 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43c8c24000, obj=) at src/switch_core_session.c:1634 #17 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #18 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #19 0x0000000000000000 in ?? () ------------------------------------------------------------------------------ (gdb) bt full #0 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 status = SWITCH_STATUS_BREAK local_fh = {file_interface = 0x1ebf798, flags = 14, fd = 0x0, samples = 0, samplerate = 8000, native_rate = 8000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 169280, speed = 0, memory_pool = 0x0, prebuf = 0, interval = 0, private_info = 0x7f43cd5595d0, handler = 0x0, pos = 0, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 30, silence_hits = 250, offset_pos = 0, samples_in = 0, samples_out = 169280, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x7f43cdd9b0e8 "\370\377\370\377\b", pre_buffer_datalen = 65536, file = 0x7f43e4a40bc5 "src/switch_ivr_play_say.c", func = 0x7f43e4a413d0 "switch_ivr_record_file", line = 518, file_path = 0x7f43cd559570 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", spool_path = 0x0, prefix = 0x7f43ccf04fc0 "/usr/local/freeswitch/sounds", max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} __func__ = "recordFile" #1 0x00007f43d243b826 in _wrap_CoreSession_recordFile (L=0x7f43cc25d370) at mod_lua_wrap.cpp:4947 arg2 = 0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav" arg4 = 30 SWIG_arg = -1 arg1 = 0x7f43dd74e860 arg3 = 30 arg5 = result = #2 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43cc25d370, func=0x7f43cce39e80, nresults=nresults at entry=0) at ldo.c:319 ci = n = cl = funcr = #3 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43cc25d370, nexeccalls=nexeccalls at entry=1) at lvm.c:587 b = nresults = 0 i = ra = cl = 0x7f43de975570 base = k = 0x7f43dcd02e20 pc = 0x7f43dcbdcd2c #4 0x00007f43d244538d in luaD_call (L=0x7f43cc25d370, func=0x7f43ccd1a6d0, nResults=) at ldo.c:377 No locals. #5 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43cc25d370, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0a90370) at ldo.c:116 lj = {previous = 0x0, b = {{__jmpbuf = {139929164567408, -7100896249400093559, 1, 139929165732864, 139929165732864, 0, 7202149886429140105, 7202145254174135433}, __mask_was_saved = 0, __saved_mask = {__val = {40, 14754083699, 80, 14653814306, 7234318537360108649, 139929164567592, 139929430523936, 40, 139929164567408, 0, 0, 139929548172288, 40, 139929164567592, 40, 0}}}}, status = 0} #6 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43cc25d370, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0a90370, old_top=, ef=) at ldo.c:463 status = oldnCcalls = 0 old_ci = 0 old_allowhooks = 1 '\001' old_errfunc = 0 #7 0x00007f43d2442be1 in lua_pcall (L=0x7f43cc25d370, nargs=0, nresults=0, errfunc=) at lapi.c:821 c = {func = 0x7f43ccd1a6d0, nresults = 0} status = func = #8 0x00007f43d24277d6 in docall (L=0x7f43cc25d370, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 status = base = 1 #9 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43cc25d370, input_code=, input_code at entry=0x7f43dc513c10 "vm_check_msgs.lua") at mod_lua.cpp:195 file = fdup = 0x7f43dd15be60 "/usr/local/freeswitch/scripts/vm_check_msgs.lua" args = error = __func__ = "lua_parse_and_execute" __PRETTY_FUNCTION__ = "int lua_parse_and_execute(lua_State*, char*)" #10 0x00007f43d24288ed in lua_function (session=0x7f43ccd3c5b8, data=) at mod_lua.cpp:476 L = mycmd = 0x7f43dc513c10 "vm_check_msgs.lua" #11 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43ccd3c5b8, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584") at src/switch_core_session.c:2809 log = lp = event = 0x0 var = channel = 0x7f43ccd421a0 expanded = 0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584" app = 0x7f43d245bc1d "lua" app_uuid_var = msg = {from = 0x7f43e4a3398e "src/switch_core_session.c", message_id = SWITCH_MESSAGE_INDICATE_APPLICATION_EXEC, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x7f43d245bc1d "lua", 0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} delim = scope = uuid_str = "262a6b64-03cc-4bdf-b374-264e20e85380\000\000\000\000\300\275?\315C\177\000\000\n", '\000' "\340, \223L\314C\177\000\000\034+\003\343C\177\000\000\060\000\000\000\000\000\000\000i\246\242\344C\177\000\000`\336\063\343C\177\000\000\n\000\000\000\000\000\000\000p\023\017\314C\177\000\000\000\000\000\000\000\000\000\000??\223\003\000\000\000\000\000,\003\343C\177\000\000p\023\017\314C\177\000\000i\246\242\344C\177\000\000\n\000\000\000\000\000\000\000a\347\213\344C\177\000\000\360j\351\001\000\000\000\000x\325\230\344C\177\000\000i\246\242\344C\177\000\000\230\023\311\001\000\000\000\000\020\344\061\326C\177\000\000\004\000\000\000\000\000\000\000\020\344\061\326C\177\000\000\000\000\000\000\000\000\000\000(\343"... app_uuid = 0x7f43d0a90a10 "262a6b64-03cc-4bdf-b374-264e20e85380" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #12 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43ccd3c5b8, app=0x7f43d631e410 "lua", arg=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", flags=flags at entry=0x0) at src/switch_core_session.c:2684 application_interface = 0x7f43dc1c9d60 status = SWITCH_STATUS_SUCCESS __func__ = "switch_core_session_execute_application_get_flags" #13 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:230 current_application = extension = 0x7f43d631e328 uuid = #14 switch_core_session_run (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:481 global_proceed = 1 index = proceed = do_extra_handlers = 1 ptr = rstatus = state = midstate = CS_EXECUTE endstate = endpoint_interface = driver_state_handler = 0x7f43dbdef600 application_state_handler = new_loops = 500 __PRETTY_FUNCTION__ = "switch_core_session_run" __func__ = "switch_core_session_run" #15 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43ccd3c5b8) at src/switch_core_session.c:1542 session = 0x7f43ccd3c5b8 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #16 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43c8c24000, obj=) at src/switch_core_session.c:1634 td = 0x7f43cc9077a8 check_status = node = pool = 0x7f43c8c23da8 pop = 0x7f43cc9077a8 check = 0 __func__ = "switch_core_session_thread_pool_worker" #17 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #18 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #19 0x0000000000000000 in ?? () No symbol table info available. ------------------------------------------------------------------------------ (gdb) thread apply all bt Thread 37 (Thread 0x7f43d0874700 (LWP 13327)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d0873e60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccec73a0, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 36 (Thread 0x7f43d08b0700 (LWP 13346)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d08afe60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccceb430, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 35 (Thread 0x7f43d0928700 (LWP 13344)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d0927e60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccb48080, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 34 (Thread 0x7f43e09e7700 (LWP 10504)): #0 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e498234d in apr_thread_cond_timedwait (cond=0x7f43dc1012e8, mutex=0x7f43dc101298, timeout=500000) at locks/unix/thread_cond.c:89 #2 0x00007f43e4979071 in apr_queue_pop_timeout (queue=0x7f43dc101258, data=0x7f43e09e6e70, timeout=) at misc/apr_queue.c:339 #3 0x00007f43e48bf895 in switch_queue_pop_timeout (queue=, data=, timeout=) at src/switch_apr.c:1061 #4 0x00007f43e48fd8d1 in switch_scheduler_task_thread (thread=, obj=) at src/switch_scheduler.c:188 #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #7 0x0000000000000000 in ?? () Thread 33 (Thread 0x7f43d85d3700 (LWP 10783)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43d8a125eb in timer_thread_run (thread=, obj=) at mod_spandsp_fax.c:211 #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #4 0x0000000000000000 in ?? () Thread 32 (Thread 0x7f43e1cfc700 (LWP 25301)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43e1cfbe60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 31 (Thread 0x7f43d8157700 (LWP 10785)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8155a60) at src/switch_time.c:683 #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8155a60) at src/switch_core_timer.c:74 #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d405b518) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 30 (Thread 0x7f43da710700 (LWP 10782)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 #2 0x00007f43da720c0a in node_thread_run (thread=, obj=) at /usr/local/src/freeswitch/src/mod/applications/mod_fifo/mod_fifo.c:2047 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #5 0x0000000000000000 in ?? () Thread 29 (Thread 0x7f43e1e42700 (LWP 10494)): #0 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e498234d in apr_thread_cond_timedwait (cond=0x7f43e5350e98, mutex=0x7f43e5350ef0, timeout=timeout at entry=10000) at locks/unix/thread_cond.c:89 #2 0x00007f43e48beb39 in switch_thread_cond_timedwait (cond=, mutex=, timeout=timeout at entry=10000) at src/switch_apr.c:380 #3 0x00007f43e48ec398 in switch_core_session_thread_pool_manager (thread=, obj=) at src/switch_core_session.c:1788 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 28 (Thread 0x7f43e09ab700 (LWP 10506)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43e09aae60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 27 (Thread 0x7f43db7eb700 (LWP 10542)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 #2 0x00007f43dba60834 in sofia_profile_worker_thread_run (thread=, obj=0x7f43dc126950) at sofia.c:2306 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #5 0x0000000000000000 in ?? () Thread 26 (Thread 0x7f43d0a54700 (LWP 25303)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0a53e60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 25 (Thread 0x7f43d8193700 (LWP 10784)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8191a60) at src/switch_time.c:683 #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8191a60) at src/switch_core_timer.c:74 #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d40573b8) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 24 (Thread 0x7f43d80df700 (LWP 10787)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d80dda60) at src/switch_time.c:683 #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d80dda60) at src/switch_core_timer.c:74 #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d4063978) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 23 (Thread 0x7f43d09dc700 (LWP 13345)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d09dbe60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43cd522460, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 22 (Thread 0x7f43d0b84700 (LWP 10790)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1c92c38, data=0x7f43d0b83e70) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e4904f3b in chat_thread_run (thread=, obj=0x1c92c38) at src/switch_loadable_module.c:680 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 21 (Thread 0x7f43e0047700 (LWP 10778)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1cb2cf8, data=0x7f43e0046e48) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dbaa5cfb in sofia_presence_event_thread_run (thread=, obj=) at sofia_presence.c:1615 #4 sofia_presence_event_thread_run (thread=, obj=) at sofia_presence.c:1592 #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #7 0x0000000000000000 in ?? () Thread 20 (Thread 0x7f43d0b48700 (LWP 10791)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43dc1da020, data=0x7f43d0b47e70) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e4904f3b in chat_thread_run (thread=, obj=0x7f43dc1da020) at src/switch_loadable_module.c:680 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 19 (Thread 0x7f43db827700 (LWP 10541)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x1ce1500) at src/switch_core_sqldb.c:1894 #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #4 0x0000000000000000 in ?? () Thread 18 (Thread 0x7f43d07d9700 (LWP 13115)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d07d8e60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43c83c23d0, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 17 (Thread 0x7f43da6d4700 (LWP 10781)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x7f43d4014360) at src/switch_core_sqldb.c:1894 #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #4 0x0000000000000000 in ?? () Thread 16 (Thread 0x7f43d8067700 (LWP 10789)): #0 0x00007f43e3e0b3cd in accept () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4987066 in apr_socket_accept (new=new at entry=0x7f43d8066e00, sock=0x1f72320, connection_context=0x7f43c86ea588) at network_io/unix/sockets.c:191 #2 0x00007f43e48bf0d5 in switch_socket_accept (new_sock=new_sock at entry=0x7f43d8066e00, sock=, pool=) at src/switch_apr.c:710 #3 0x00007f43dbdf9018 in mod_event_socket_runtime () at /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2835 #4 0x00007f43e4900df3 in switch_loadable_module_exec (thread=0x1c92bb8, obj=0x1c92798) at src/switch_loadable_module.c:98 #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #7 0x0000000000000000 in ?? () Thread 15 (Thread 0x7f43d0a18700 (LWP 25302)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0a17e60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 14 (Thread 0x7f43e53b6760 (LWP 10491)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 #2 0x00007f43e48f7d9a in switch_core_runtime_loop (bg=0, bg at entry=1) at src/switch_core.c:994 #3 0x0000000000402cae in main (argc=, argv=) at src/switch.c:1184 Thread 13 (Thread 0x7f43d0ad0700 (LWP 25300)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0acfe60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 12 (Thread 0x7f43d0b0c700 (LWP 10020)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495ae4c in timer_next (timer=0x7f43cd5785e8) at src/switch_time.c:683 #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43cd5785e8) at src/switch_core_timer.c:74 #3 0x00007f43e491969d in rtp_common_read (rtp_session=rtp_session at entry=0x7f43cd568258, payload_type=payload_type at entry=0x7f43dccac4f8 "", flags=flags at entry=0x7f43dccac514, io_flags=io_flags at entry=2) at src/switch_rtp.c:3423 #4 0x00007f43e491ab83 in switch_rtp_zerocopy_read_frame (rtp_session=0x7f43cd568258, frame=frame at entry=0x7f43dccac4b8, io_flags=io_flags at entry=2) at src/switch_rtp.c:4115 #5 0x00007f43dba4c90f in sofia_read_frame (session=0x7f43dd344798, frame=0x7f43d0b0a848, flags=2, stream_id=) at mod_sofia.c:1118 #6 0x00007f43e48f182e in switch_core_session_read_frame (session=session at entry=0x7f43dd344798, frame=frame at entry=0x7f43d0b0a848, flags=flags at entry=2, stream_id=stream_id at entry=0) at src/switch_core_io.c:224 #7 0x00007f43e493b9bd in switch_ivr_play_file (session=0x7f43dd344798, fh=0x7f43d0b0af60, file=0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", args=0x7f43dc61fc18) at src/switch_ivr_play_say.c:1626 #8 0x00007f43e4974e09 in CoreSession::streamFile (this=0x7f43dc61fc10, file=0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", starting_sample_count=) at src/switch_cpp.cpp:967 #9 0x00007f43d24396ae in _wrap_CoreSession_streamFile (L=0x7f43d48aa040) at mod_lua_wrap.cpp:5578 #10 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43d48aa040, func=0x7f43d43852b0, nresults=nresults at entry=0) at ldo.c:319 #11 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43d48aa040, nexeccalls=2, nexeccalls at entry=1) at lvm.c:587 #12 0x00007f43d244538d in luaD_call (L=0x7f43d48aa040, func=0x7f43d47087f0, nResults=) at ldo.c:377 #13 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43d48aa040, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0b0b370) at ldo.c:116 #14 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43d48aa040, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0b0b370, old_top=, ef=) at ldo.c:463 #15 0x00007f43d2442be1 in lua_pcall (L=0x7f43d48aa040, nargs=0, nresults=0, errfunc=) at lapi.c:821 #16 0x00007f43d24277d6 in docall (L=0x7f43d48aa040, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 #17 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43d48aa040, input_code=, input_code at entry=0x7f43dc7536a0 "vm_check_msgs.lua") at mod_lua.cpp:195 #18 0x00007f43d24288ed in lua_function (session=0x7f43dd344798, data=) at mod_lua.cpp:476 #19 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43dd344798, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412") at src/switch_core_session.c:2809 #20 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43dd344798, app=0x7f43cce4fd40 "lua", arg=0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412", flags=flags at entry=0x0) at src/switch_core_session.c:2684 #21 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43dd344798) at src/switch_core_state_machine.c:230 #22 switch_core_session_run (session=0x7f43dd344798) at src/switch_core_state_machine.c:481 #23 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43dd344798) at src/switch_core_session.c:1542 #24 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43cce85d80, obj=) at src/switch_core_session.c:1634 #25 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #26 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #27 0x0000000000000000 in ?? () Thread 11 (Thread 0x7f43e52ee700 (LWP 10493)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 #2 0x00007f43e48d9905 in pool_thread (thread=, obj=) at src/switch_core_memory.c:565 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #5 0x0000000000000000 in ?? () Thread 10 (Thread 0x7f43d09a0700 (LWP 13119)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d099fe60) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43cd9170d0, obj=) at src/switch_core_session.c:1615 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 9 (Thread 0x7f43e1cc0700 (LWP 10498)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e1e05e70, data=0x7f43e1cbfe70) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e494ffb3 in log_thread (t=, obj=) at src/switch_log.c:294 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 8 (Thread 0x7f43d811b700 (LWP 10786)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8119a60) at src/switch_time.c:683 #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8119a60) at src/switch_core_timer.c:74 #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d405f748) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 7 (Thread 0x7f43e00c1700 (LWP 10507)): #0 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43dbb51e19 in su_epoll_port_wait_events (self=0x1c9e950, tout=) at su_epoll_port.c:495 #2 0x00007f43dbb513f3 in su_base_port_step (self=0x1c9e950, tout=1000) at su_base_port.c:467 #3 0x00007f43dba73ddb in sofia_profile_thread_run (thread=, obj=0x7f43dc126950) at sofia.c:2664 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 6 (Thread 0x7f43e0c0e700 (LWP 10503)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 #2 0x00007f43e48e59cd in switch_core_sql_db_thread (thread=, obj=) at src/switch_core_sqldb.c:1238 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #5 0x0000000000000000 in ?? () Thread 5 (Thread 0x7f43d80a3700 (LWP 10788)): #0 0x00007f43e3e0b18d in read () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e495b822 in softtimer_runtime () at src/switch_time.c:940 #2 0x00007f43e4900df3 in switch_loadable_module_exec (thread=0x1c92700, obj=0x1c921e0) at src/switch_loadable_module.c:98 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #5 0x0000000000000000 in ?? () Thread 4 (Thread 0x7f43e0084700 (LWP 10540)): #0 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 #1 0x00007f43dbb51e19 in su_epoll_port_wait_events (self=0x1ccfe80, tout=) at su_epoll_port.c:495 #2 0x00007f43dbb5128d in su_base_port_run (self=0x1ccfe80) at su_base_port.c:349 #3 0x00007f43dbb561a8 in su_pthread_port_clone_main (varg=0x7f43e00c07d0) at su_pthread_port.c:343 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 3 (Thread 0x7f43e10d5700 (LWP 10502)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x7f43dc0fb170) at src/switch_core_sqldb.c:1894 #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #4 0x0000000000000000 in ?? () Thread 2 (Thread 0x7f43e1da3700 (LWP 10495)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1b40880, data=0x7f43e1da2e50) at misc/apr_queue.c:276 #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 #3 0x00007f43e490d889 in switch_event_dispatch_thread (thread=, obj=0x1b40880) at src/switch_event.c:305 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #6 0x0000000000000000 in ?? () Thread 1 (Thread 0x7f43d0a91700 (LWP 12924)): #0 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 #1 0x00007f43d243b826 in _wrap_CoreSession_recordFile (L=0x7f43cc25d370) at mod_lua_wrap.cpp:4947 #2 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43cc25d370, func=0x7f43cce39e80, nresults=nresults at entry=0) at ldo.c:319 #3 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43cc25d370, nexeccalls=nexeccalls at entry=1) at lvm.c:587 #4 0x00007f43d244538d in luaD_call (L=0x7f43cc25d370, func=0x7f43ccd1a6d0, nResults=) at ldo.c:377 #5 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43cc25d370, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0a90370) at ldo.c:116 #6 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43cc25d370, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0a90370, old_top=, ef=) at ldo.c:463 #7 0x00007f43d2442be1 in lua_pcall (L=0x7f43cc25d370, nargs=0, nresults=0, errfunc=) at lapi.c:821 #8 0x00007f43d24277d6 in docall (L=0x7f43cc25d370, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 #9 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43cc25d370, input_code=, input_code at entry=0x7f43dc513c10 "vm_check_msgs.lua") at mod_lua.cpp:195 #10 0x00007f43d24288ed in lua_function (session=0x7f43ccd3c5b8, data=) at mod_lua.cpp:476 #11 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43ccd3c5b8, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584") at src/switch_core_session.c:2809 #12 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43ccd3c5b8, app=0x7f43d631e410 "lua", arg=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", flags=flags at entry=0x0) at src/switch_core_session.c:2684 #13 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:230 #14 switch_core_session_run (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:481 #15 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43ccd3c5b8) at src/switch_core_session.c:1542 #16 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43c8c24000, obj=) at src/switch_core_session.c:1634 #17 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 #18 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 #19 0x0000000000000000 in ?? () ------------------------------------------------------------------------------ (gdb) thread apply all bt full Thread 37 (Thread 0x7f43d0874700 (LWP 13327)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d0873e60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccec73a0, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43ccec7148 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 36 (Thread 0x7f43d08b0700 (LWP 13346)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d08afe60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccceb430, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43ccceb1d8 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 35 (Thread 0x7f43d0928700 (LWP 13344)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d0927e60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43ccb48080, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43ccb47e28 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 34 (Thread 0x7f43e09e7700 (LWP 10504)): #0 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e498234d in apr_thread_cond_timedwait (cond=0x7f43dc1012e8, mutex=0x7f43dc101298, timeout=500000) at locks/unix/thread_cond.c:89 rv = then = abstime = {tv_sec = 1415641366, tv_nsec = 792983000} #2 0x00007f43e4979071 in apr_queue_pop_timeout (queue=0x7f43dc101258, data=0x7f43e09e6e70, timeout=) at misc/apr_queue.c:339 rv = 0 #3 0x00007f43e48bf895 in switch_queue_pop_timeout (queue=, data=, timeout=) at src/switch_apr.c:1061 No locals. #4 0x00007f43e48fd8d1 in switch_scheduler_task_thread (thread=, obj=) at src/switch_scheduler.c:188 pop = 0x7f43c8002a60 __func__ = "switch_scheduler_task_thread" #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #7 0x0000000000000000 in ?? () No symbol table info available. Thread 33 (Thread 0x7f43d85d3700 (LWP 10783)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43d8a125eb in timer_thread_run (thread=, obj=) at mod_spandsp_fax.c:211 timer = {interval = 20, flags = 1, samples = 160, samplecount = 160, timer_interface = 0x1c93450, memory_pool = 0x1eb5038, private_info = 0x1eb5120, diff = 0, tick = 0} pvt = __func__ = "timer_thread_run" #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #4 0x0000000000000000 in ?? () No symbol table info available. Thread 32 (Thread 0x7f43e1cfc700 (LWP 25301)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43e1cfbe60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 pop = 0x7f43cc710210 q = 0x7f43e0123ad8 my_id = __func__ = "sofia_msg_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 31 (Thread 0x7f43d8157700 (LWP 10785)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8155a60) at src/switch_time.c:683 private_info = 0x7f43d6567180 cond_index = delta = #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8155a60) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d405b518) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 is_open = use_fh = 0x7f43d8155b20 olen = 160 abuf = "G\377\215\377\335\377&\000W\000p\000h\000S\000U\000b\000y\000\211\000m\000)\000\313\377b\377\b\377\312\376\265\376\325\376\035\377|\377\341\377\062\000e\000\210\000\233\000\247\000\303\000\331\000\314\000\244\000_\000\361\377l\377\362\376\243\376\214\376\243\376\336\376=\377\267\377(\000\203\000\301\000\324\000\334\000\347\000\350\000\335\000\272\000r\000\004\000\202\377\004\377\236\376c\376]\376\227\376\004\377\207\377\006\000g\000\255\000\341\000\003\001\031\001#\001*\001#\001\371\000\255\000F\000\332\377|\377-\377\005\377\020\377>\377\211\377\347\377<\000z\000\244\000\302\000\327\000\362\000\016\001\035\001\031\001\370\000\266\000X\000\341\377i\377\024\377\365\376\v\377@\377\202\377\307\377\n\000D\000j\000\222\000\317\000\022\001P\001x\001x\001D\001\336\000^\000\335\377n\377\034\377\367\376\006\377\065\377z\377\300\377\364\377 \000Q\000\213\000\315\000\a\001+\001-\001\004\001\261\000\067\000\253\377+\377\323\376\262\376\277\376\361\376>\377\223\377\347\377\071\000~\000\265\000\354\000)\001V\001\\\001\070\001\361\000\214\000\021\000\225\377\060\377\361\376\355\376\"\377q\377\275\377\375\377", '\000' fname = 0x7f43d8155c50 "/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav" source = 0x7f43d405b518 fh = {file_interface = 0x1ebf798, flags = 2061, fd = 0x0, samples = 3052403, samplerate = 8000, native_rate = 8000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 3972903907, speed = 0, memory_pool = 0x7f43d4b75f48, prebuf = 65536, interval = 0, private_info = 0x7f43d4b76088, handler = 0x0, pos = 3972903907, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, samples_in = 2064384, samples_out = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x7f43d41c9020, pre_buffer_data = 0x7f43d53a7958 "*\002U\002y\002\177\002l\002B\002\001\002\256\001I", , pre_buffer_datalen = 65536, file = 0x7f43d2875c48 "/usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c", func = 0x7f43d2875e80 "read_stream_thread", line = 213, file_path = 0x7f43d4b76030 "/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav", spool_path = 0x0, prefix = 0x0, max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} cp = file_buf = "danza-espanola-op-37-h-142-xii-arabesca.wav\000wav", '\000' path_buf = "/usr/local/freeswitch/sounds/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav\000wav", '\000' timer = {interval = 20, flags = 0, samples = 160, samplecount = 2000640, timer_interface = 0x1c93450, memory_pool = 0x7f43d6566e78, private_info = 0x7f43d6567180, diff = 0, tick = 0} fd = -1 audio_buffer = 0x1ec8c50 dist_buf = 0x1f0a598 "\222\376\317\375\360\375\214\375d\375\246\376\337\377x" used = skip = 0 temp_pool = 0x7f43d6566e78 __func__ = "read_stream_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 30 (Thread 0x7f43da710700 (LWP 10782)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 87588} #2 0x00007f43da720c0a in node_thread_run (thread=, obj=) at /usr/local/src/freeswitch/src/mod/applications/mod_fifo/mod_fifo.c:2047 ppl_waiting = consumer_total = idle_consumers = found = node = 0x0 last = this_node = cur_priority = 1 __func__ = "node_thread_run" #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. Thread 29 (Thread 0x7f43e1e42700 (LWP 10494)): #0 0x00007f43e3e0864b in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e498234d in apr_thread_cond_timedwait (cond=0x7f43e5350e98, mutex=0x7f43e5350ef0, timeout=timeout at entry=10000) at locks/unix/thread_cond.c:89 rv = then = abstime = {tv_sec = 1415641366, tv_nsec = 610472000} #2 0x00007f43e48beb39 in switch_thread_cond_timedwait (cond=, mutex=, timeout=timeout at entry=10000) at src/switch_apr.c:380 st = #3 0x00007f43e48ec398 in switch_core_session_thread_pool_manager (thread=, obj=) at src/switch_core_session.c:1788 check = 1 ttl = 0 xsleep = sleep = next = 1415641376589480 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 28 (Thread 0x7f43e09ab700 (LWP 10506)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43e09aae60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 pop = 0x200eb90 q = 0x7f43e0123ad8 my_id = __func__ = "sofia_msg_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 27 (Thread 0x7f43db7eb700 (LWP 10542)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 64434} #2 0x00007f43dba60834 in sofia_profile_worker_thread_run (thread=, obj=0x7f43dc126950) at sofia.c:2306 profile = 0x7f43dc126950 ireg_loops = 12 gateway_loops = __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" __func__ = "sofia_profile_worker_thread_run" #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. Thread 26 (Thread 0x7f43d0a54700 (LWP 25303)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0a53e60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 pop = 0x7f43d426c480 q = 0x7f43e0123ad8 my_id = __func__ = "sofia_msg_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 25 (Thread 0x7f43d8193700 (LWP 10784)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8191a60) at src/switch_time.c:683 private_info = 0x7f43d59c0de0 cond_index = delta = #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8191a60) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d40573b8) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 is_open = use_fh = 0x7f43d8191b20 olen = 160 abuf = "\005\000\003\000\000\000\000\000\001\000\004\000\b\000\v\000\f\000\n\000\003\000\376\377\376\377\000\000\005\000\b\000\006\000\002\000\000\000\375\377\373\377\376\377\002\000\004\000\004\000\001\000\376\377\375\377\375\377\377\377\003\000\005\000\a\000\005\000\377\377\375\377\373\377\373\377\000\000\004\000\005\000\004\000\001\000\376\377\374\377\374\377\375\377\001\000\003\000\002\000\377\377\374\377\374\377\377\377\001\000\003\000\005\000\003\000\377\377\370\377\365\377\370\377\372\377\377\377\004\000\004\000\000\000\372\377\370\377\370\377\371\377\375\377\002\000\002\000\376\377\372\377\370\377\367\377\374\377\000\000\000\000\001\000\377\377\372\377\370\377\372\377\376\377\003\000\003\000\003\000\000\000\372\377\366\377\365\377\367\377\371\377\374\377\376\377\374\377\371\377\367\377\370\377\372\377\377\377\003\000\004\000\003\000\000\000\374\377\375\377\002\000\a\000\n\000\f\000\b\000\002\000\375\377\375\377\000\000\004\000\005\000\005\000\002\000\374\377\371\377\371\377\372\377\376\377\003\000\003\000\000\000\375\377\372\377\372\377\376\377\001\000\003\000\005\000\002\000\376\377\372\377\372\377\376\377\001\000\005\000\t\000\a\000\002\000\376\377\375\377\377\377\003\000\006\000\t\000\b\000\005\000\001\000\377\377\376\377\001", '\000' fname = 0x7f43d8191c50 "/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav" source = 0x7f43d40573b8 fh = {file_interface = 0x1ebf798, flags = 2061, fd = 0x0, samples = 3204726, samplerate = 8000, native_rate = 8000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 3973384203, speed = 0, memory_pool = 0x7f43d5cf5f68, prebuf = 65536, interval = 0, private_info = 0x7f43d5cf60a0, handler = 0x0, pos = 3973384203, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, samples_in = 884736, samples_out = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x7f43d46d0090, pre_buffer_data = 0x7f43d5f9ebc8 "\373\377", pre_buffer_datalen = 65536, file = 0x7f43d2875c48 "/usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c", func = 0x7f43d2875e80 "read_stream_thread", line = 213, file_path = 0x7f43d5cf6050 "/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav", spool_path = 0x0, prefix = 0x0, max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} cp = file_buf = "suite-espanola-op-47-leyenda.wav\000\061-preludio.wav", '\000' path_buf = "/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav\000\061-preludio.wav", '\000' timer = {interval = 20, flags = 0, samples = 160, samplecount = 840320, timer_interface = 0x1c93450, memory_pool = 0x7f43d59c0b18, private_info = 0x7f43d59c0de0, diff = 0, tick = 0} fd = -1 audio_buffer = 0x1e98f50 dist_buf = 0x1ef9588 "\354\001\375\001\262\001]", used = skip = 0 temp_pool = 0x7f43d59c0b18 __func__ = "read_stream_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 24 (Thread 0x7f43d80df700 (LWP 10787)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d80dda60) at src/switch_time.c:683 private_info = 0x7f43d4b55648 cond_index = delta = #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d80dda60) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d4063978) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 is_open = use_fh = 0x7f43d80ddb20 olen = 640 abuf = ";\377\261\376(\376\260\375\060\375\254\374?\374\351\373\240\373f\373\060\373\v\373\f\373\n\373\350\372\257\372`\372)\372*\372\063\372\062\372Z\372\253\372\374\372`\373\324\373F\374\324\374y\375\024\376\251\376#\377\207\377\367\377u\000\370\000\202\001\032\002\272\002G\003\252\003\370\003Q\004\277\004\066\005\244\005\355\005\366\005\317\005\200\005\377\004v\004\v\004\261\003`\003\001\003{\002\374\001\247\001[\001\030\001\345\000\245\000c\000\035\000\251\377\017\377\200\376\v\376\254\375o\375J\375\"\375\357\374\264\374~\374[\374D\374)\374\002\374\326\373\261\373\220\373i\373D\373)\373\037\373E\373\247\373\025\374\201\374\372\374y\375\a\376\256\376X\377\357\377u\000\344\000\071\001\215\001\351\001A\002\220\002\315\002\003\003M\003\227\003\313\003\375\003%\004.\004(\004\037\004\370\003\246\003\063\003\265\002H\002\351\001\212\001-\001\317\000j\000\005\000\246\377N\377\005\377\304\376u\376!\376\317\375q\375\r\375\264\374m\374B\374\"\374\373\373\347\373\361\373\363\373\337\373\302\373\246\373\242\373\301\373\342\373\352\373\341\373\323\373\321\373\374\373R\374\272\374L\375\375\375\232\376'\377\267\377O\000\353\000\201\001\025\002\254\002P\003\366\003}\004\326\004\r\005B\005~\005\266\005\346\005\t\006)\006E\006\061\006\351\005\230\005K\005\b\005\331\004\247\004\\\004\001\004\242\003\062\003\277\002a\002\t\002\256\001M\001\324\000Z\000\361\377\202\377\017\377\250\376?\376\326\375\204\375C\375\005\375\313\374\227\374j\374=\374\b\374"... fname = 0x7f43d80ddc50 "/usr/local/freeswitch/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav" source = 0x7f43d4063978 fh = {file_interface = 0x1ebf798, flags = 2061, fd = 0x0, samples = 8414169, samplerate = 32000, native_rate = 32000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 16051800762, speed = 0, memory_pool = 0x7f43d5cf81c8, prebuf = 65536, interval = 0, private_info = 0x7f43d5cf8310, handler = 0x0, pos = 16051800762, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, samples_in = 196608, samples_out = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x7f43d4abe010, pre_buffer_data = 0x7f43d4fbd008 "V\a\226\a\267\a\251\ag\a\361\006V\006\262\005\027\005\210\004\v\004\242\003J\003\351\002k\002\320\001$\001z", pre_buffer_datalen = 65536, file = 0x7f43d2875c48 "/usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c", func = 0x7f43d2875e80 "read_stream_thread", line = 213, file_path = 0x7f43d5cf82b0 "/usr/local/freeswitch/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav", spool_path = 0x0, prefix = 0x0, max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} cp = file_buf = "partita-no-3-in-e-major-bwv-1006-1-preludio.wav", '\000' path_buf = "/usr/local/freeswitch/sounds/music/32000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav", '\000' timer = {interval = 20, flags = 0, samples = 640, samplecount = 147840, timer_interface = 0x1c93450, memory_pool = 0x7f43d4b552e8, private_info = 0x7f43d4b55648, diff = 0, tick = 0} fd = -1 audio_buffer = 0x1ec19d0 dist_buf = 0x1f56648 "\202\375\210\375\220\375\236\375\257\375\303\375\340\375\377\375\035\376?\376c\376\210\376\257\376\323\376\370\376\036\377B\377g\377\211\377\246\377\305\377\353\377\021" used = skip = 0 temp_pool = 0x7f43d4b552e8 __func__ = "read_stream_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 23 (Thread 0x7f43d09dc700 (LWP 13345)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d09dbe60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43cd522460, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43cd522208 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 22 (Thread 0x7f43d0b84700 (LWP 10790)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1c92c38, data=0x7f43d0b83e70) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e4904f3b in chat_thread_run (thread=, obj=0x1c92c38) at src/switch_loadable_module.c:680 pop = 0x0 q = 0x1c92c38 __func__ = "chat_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 21 (Thread 0x7f43e0047700 (LWP 10778)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1cb2cf8, data=0x7f43e0046e48) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dbaa5cfb in sofia_presence_event_thread_run (thread=, obj=) at sofia_presence.c:1615 count = 0 pop = 0x7f43cc3e8bc0 #4 sofia_presence_event_thread_run (thread=, obj=) at sofia_presence.c:1592 done = 0 #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #7 0x0000000000000000 in ?? () No symbol table info available. Thread 20 (Thread 0x7f43d0b48700 (LWP 10791)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43dc1da020, data=0x7f43d0b47e70) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e4904f3b in chat_thread_run (thread=, obj=0x7f43dc1da020) at src/switch_loadable_module.c:680 pop = 0x0 q = 0x7f43dc1da020 __func__ = "chat_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 19 (Thread 0x7f43db827700 (LWP 10541)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x1ce1500) at src/switch_core_sqldb.c:1894 i = lc = written = iterations = qm = 0x1ce1500 i = __func__ = "switch_user_sql_thread" #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #4 0x0000000000000000 in ?? () No symbol table info available. Thread 18 (Thread 0x7f43d07d9700 (LWP 13115)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d07d8e60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43c83c23d0, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43c83c2178 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 17 (Thread 0x7f43da6d4700 (LWP 10781)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x7f43d4014360) at src/switch_core_sqldb.c:1894 i = lc = written = iterations = qm = 0x7f43d4014360 i = __func__ = "switch_user_sql_thread" #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #4 0x0000000000000000 in ?? () No symbol table info available. Thread 16 (Thread 0x7f43d8067700 (LWP 10789)): #0 0x00007f43e3e0b3cd in accept () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4987066 in apr_socket_accept (new=new at entry=0x7f43d8066e00, sock=0x1f72320, connection_context=0x7f43c86ea588) at network_io/unix/sockets.c:191 No locals. #2 0x00007f43e48bf0d5 in switch_socket_accept (new_sock=new_sock at entry=0x7f43d8066e00, sock=, pool=) at src/switch_apr.c:710 No locals. #3 0x00007f43dbdf9018 in mod_event_socket_runtime () at /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2835 pool = 0x1f72128 listener_pool = 0x7f43c86ea588 rv = sa = 0x1f72250 inbound_socket = 0x7f43c86ea6b0 listener = 0x7f43c830ebb0 x = 0 errs = 0 __func__ = "mod_event_socket_runtime" #4 0x00007f43e4900df3 in switch_loadable_module_exec (thread=0x1c92bb8, obj=0x1c92798) at src/switch_loadable_module.c:98 status = ts = 0x1c92798 module = 0x1cb0f90 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #5 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #6 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #7 0x0000000000000000 in ?? () No symbol table info available. Thread 15 (Thread 0x7f43d0a18700 (LWP 25302)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0a17e60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 pop = 0x7f43d426f2c0 q = 0x7f43e0123ad8 my_id = __func__ = "sofia_msg_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 14 (Thread 0x7f43e53b6760 (LWP 10491)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 343341} #2 0x00007f43e48f7d9a in switch_core_runtime_loop (bg=0, bg at entry=1) at src/switch_core.c:994 No locals. #3 0x0000000000402cae in main (argc=, argv=) at src/switch.c:1184 pid_path = "/usr/local/freeswitch/run/freeswitch.pid", '\000' pid_buffer = "10491", '\000' old_pid_buffer = "21125", '\000' pid_len = 5 old_pid_len = 5 err = 0x7f43e4a3c8c7 "Success" nf = do_wait = runas_user = runas_group = reincarnate = reincarnate_reexec = fds = {0, 0} nc = SWITCH_TRUE pid = 10491 i = x = opts = opts_str = '\000' local_argv = {0x7fffb67dde84 "/usr/local/freeswitch/bin/freeswitch", 0x7fffb67ddea9 "-u", 0x7fffb67ddeac "freeswitch", 0x7fffb67ddeb7 "-g", 0x7fffb67ddeba "freeswitch", 0x7fffb67ddec5 "-rp", 0x7fffb67ddec9 "-nc", 0x7fffb67ddecd "-nonat", 0x7fffb67dded4 "-rp", 0x7fffb67dded8 "-nonat", 0x0 } local_argc = arg_argv = {0x0 } alt_dirs = log_set = run_set = do_kill = priority = flags = ret = 0 destroy_status = fd = 0x1b2db60 pool = 0x1b2da28 Thread 13 (Thread 0x7f43d0ad0700 (LWP 25300)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e0123ad8, data=0x7f43d0acfe60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43dba8321d in sofia_msg_thread_run (thread=, obj=0x7f43e0123ad8) at sofia.c:1669 pop = 0x7f43d40cef20 q = 0x7f43e0123ad8 my_id = __func__ = "sofia_msg_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 12 (Thread 0x7f43d0b0c700 (LWP 10020)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495ae4c in timer_next (timer=0x7f43cd5785e8) at src/switch_time.c:683 private_info = 0x7f43dccae1a0 cond_index = delta = #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43cd5785e8) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #3 0x00007f43e491969d in rtp_common_read (rtp_session=rtp_session at entry=0x7f43cd568258, payload_type=payload_type at entry=0x7f43dccac4f8 "", flags=flags at entry=0x7f43dccac514, io_flags=io_flags at entry=2) at src/switch_rtp.c:3423 do_cng = 0 read_pretriggered = 0 session = 0x7f43dd344798 channel = 0x7f43dd34a380 bytes = 0 rtcp_bytes = 0 status = SWITCH_STATUS_SUCCESS poll_status = SWITCH_STATUS_SUCCESS rtcp_status = rtcp_poll_status = check = 0 ret = -1 sleep_mss = 20000 poll_loop = 0 fdr = 0 rtcp_fdr = 0 hot_socket = 0 read_loops = 0 __func__ = "rtp_common_read" #4 0x00007f43e491ab83 in switch_rtp_zerocopy_read_frame (rtp_session=0x7f43cd568258, frame=frame at entry=0x7f43dccac4b8, io_flags=io_flags at entry=2) at src/switch_rtp.c:4115 bytes = 0 #5 0x00007f43dba4c90f in sofia_read_frame (session=0x7f43dd344798, frame=0x7f43d0b0a848, flags=2, stream_id=) at mod_sofia.c:1118 status = tech_pvt = 0x7f43dccac458 channel = sanity = rtcp_frame = {report_count = 0, packet_type = 0, ssrc = 0, ntp_msw = 0, ntp_lsw = 0, timestamp = 0, packet_count = 0, octect_count = 0, nb_reports = 0, reports = {{ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}, {ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}, {ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}, {ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}, {ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}}} __PRETTY_FUNCTION__ = "sofia_read_frame" __func__ = "sofia_read_frame" #6 0x00007f43e48f182e in switch_core_session_read_frame (session=session at entry=0x7f43dd344798, frame=frame at entry=0x7f43d0b0a848, flags=flags at entry=2, stream_id=stream_id at entry=0) at src/switch_core_io.c:224 ptr = status = SWITCH_STATUS_FALSE need_codec = 0 perfect = 0 do_bugs = 0 do_resample = 0 is_cng = 0 tap_only = 0 codec_impl = flag = 0 i = 2 __PRETTY_FUNCTION__ = "switch_core_session_read_frame" __func__ = "switch_core_session_read_frame" #7 0x00007f43e493b9bd in switch_ivr_play_file (session=0x7f43dd344798, fh=0x7f43d0b0af60, file=0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", args=0x7f43dc61fc18) at src/switch_ivr_play_say.c:1626 read_frame = 0x0 tstatus = do_speed = 1 last_speed = -1 f = channel = 0x7f43dd34a380 abuf = 0x309f800 dtmf = {digit = 0 '\000', duration = 0, flags = 0, source = SWITCH_DTMF_UNKNOWN} interval = samples = framelen = 320 sample_start = 0 ilen = 160 olen = 160 llen = 160 write_frame = {codec = 0x7f43d0b0a8a0, source = 0x0, packet = 0x0, packetlen = 0, extra_data = 0x0, data = 0x309f800, datalen = 320, buflen = 65536, samples = 160, rate = 8000, payload = 0 '\000', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0, user_data = 0x0} timer = {interval = 0, flags = 0, samples = 0, samplecount = 0, timer_interface = 0x0, memory_pool = 0x0, private_info = 0x0, diff = 0, tick = 0} codec = {codec_interface = 0x1ca8fd0, implementation = 0x1ca9598, fmtp_in = 0x0, fmtp_out = 0x0, flags = 259, memory_pool = 0x7f43dd605e88, private_info = 0x0, agreed_pt = 0 '\000', mutex = 0x7f43ccefbfd0, next = 0x0, session = 0x0, cur_frame = 0x0} pool = 0x7f43dd605e88 codec_name = 0x7f43e4a339a8 "L16" status = SWITCH_STATUS_SUCCESS lfh = {file_interface = 0x0, flags = 0, fd = 0x0, samples = 0, samplerate = 0, native_rate = 0, channels = 0, format = 0, sections = 0, seekable = 0, sample_count = 0, speed = 0, memory_pool = 0x0, prebuf = 0, interval = 0, private_info = 0x0, handler = 0x0, pos = 0, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, samples_in = 139929450727320, samples_out = 139929450750848, vol = 0, resampler = 0x7f43e48e8e85, buffer = 0x7f43dd34a380, dbuf = 0x7f43e48c3dc1 "\205\300t\351\213CP9CT[\017\225\300\017\266\300\303ffff.\017\037\204", dbuflen = 139929450750848, pre_buffer = 0x7f43e48c34be, pre_buffer_data = 0x7f43dd34a380 "\240\321\312\334C\177", pre_buffer_datalen = 139929450750848, file = 0x0, func = 0x7f43e49456a0 "\205\300\017\205*\003", line = 0, file_path = 0x0, spool_path = 0x0, prefix = 0x7f43c8000020 "", max_samples = -933374120, params = 0x7f43c85dd6f4, cur_channels = 3361593104, cur_samplerate = 32579} p = 0x2c02cf5 "2014-10-09 17:20:43" ext = prefix = 0x7f43ccefbfb0 "/usr/local/freeswitch/sounds" timer_name = 0x0 prebuf = alt = sleep_val = play_delimiter_val = play_delimiter = sleep_val_i = 250 eof = 0 bread = read_impl = {codec_type = SWITCH_CODEC_TYPE_AUDIO, ianacode = 0 '\000', iananame = 0x1cab138 "PCMU", fmtp = 0x0, samples_per_second = 8000, actual_samples_per_second = 8000, bits_per_second = 64000, microseconds_per_packet = 20000, samples_per_packet = 160, decoded_bytes_per_packet = 320, encoded_bytes_per_packet = 160, number_of_channels = 1 '\001', codec_frames_per_packet = 160, init = 0x7f43e495ecb0 , encode = 0x7f43e495ecc0 , decode = 0x7f43e495ef10 , destroy = 0x7f43e495ed50 , codec_id = 61, impl_id = 72, next = 0x1cab050} file_dup = argv = {0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", 0x0 } argc = 1 cur = -512 done = timeout_samples = 0 timeout_as_success = SWITCH_FALSE var = more_data = 0 event = 0x0 test_native = 0 last_native = 0 buflen = __func__ = "switch_ivr_play_file" __PRETTY_FUNCTION__ = "switch_ivr_play_file" #8 0x00007f43e4974e09 in CoreSession::streamFile (this=0x7f43dc61fc10, file=0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", starting_sample_count=) at src/switch_cpp.cpp:967 status = prebuf = local_fh = {file_interface = 0x1ebf798, flags = 264205, fd = 0x0, samples = 206720, samplerate = 8000, native_rate = 8000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 163840, speed = 0, memory_pool = 0x3f04848, prebuf = 0, interval = 0, private_info = 0x3f049a0, handler = 0x0, pos = 163840, audio_buffer = 0x2cd9400, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 157600, samples_in = 163840, samples_out = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x0, pre_buffer_datalen = 0, file = 0x7f43e4a40bc5 "src/switch_ivr_play_say.c", func = 0x7f43e4a41430 "switch_ivr_play_file", line = 1236, file_path = 0x3f04930 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav", spool_path = 0x0, prefix = 0x7f43ccefbfb0 "/usr/local/freeswitch/sounds", max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} __func__ = "streamFile" #9 0x00007f43d24396ae in _wrap_CoreSession_streamFile (L=0x7f43d48aa040) at mod_lua_wrap.cpp:5578 arg2 = 0x7f43d50f4d48 "/usr/local/freeswitch/storage/voicemail/212.225.254.22/952274412/msg_484200a8-3be4-4086-91ef-73f11e321d34.wav" SWIG_arg = -1 arg1 = 0x7f43dc61fc10 arg3 = result = #10 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43d48aa040, func=0x7f43d43852b0, nresults=nresults at entry=0) at ldo.c:319 ci = n = cl = funcr = #11 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43d48aa040, nexeccalls=2, nexeccalls at entry=1) at lvm.c:587 b = nresults = 0 i = ra = cl = 0x7f43dc20f750 base = k = 0x7f43dcd49d70 pc = 0x7f43dcafe49c #12 0x00007f43d244538d in luaD_call (L=0x7f43d48aa040, func=0x7f43d47087f0, nResults=) at ldo.c:377 No locals. #13 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43d48aa040, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0b0b370) at ldo.c:116 lj = {previous = 0x0, b = {{__jmpbuf = {139929305391168, -7100896249400093559, 1, 139929303008400, 139929303008400, 0, 7202149822541501577, 7202145254174135433}, __mask_was_saved = 0, __saved_mask = {__val = {40, 14754083699, 80, 14653814306, 7234318537360108649, 139929305391352, 139929430523936, 40, 139929305391168, 0, 0, 139929548172288, 40, 139929305391352, 40, 0}}}}, status = 0} #14 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43d48aa040, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0b0b370, old_top=, ef=) at ldo.c:463 status = oldnCcalls = 0 old_ci = 0 old_allowhooks = 1 '\001' old_errfunc = 0 #15 0x00007f43d2442be1 in lua_pcall (L=0x7f43d48aa040, nargs=0, nresults=0, errfunc=) at lapi.c:821 c = {func = 0x7f43d47087f0, nresults = 0} status = func = #16 0x00007f43d24277d6 in docall (L=0x7f43d48aa040, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 status = base = 1 #17 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43d48aa040, input_code=, input_code at entry=0x7f43dc7536a0 "vm_check_msgs.lua") at mod_lua.cpp:195 file = fdup = 0x7f43dc23c1a0 "/usr/local/freeswitch/scripts/vm_check_msgs.lua" args = error = __func__ = "lua_parse_and_execute" __PRETTY_FUNCTION__ = "int lua_parse_and_execute(lua_State*, char*)" #18 0x00007f43d24288ed in lua_function (session=0x7f43dd344798, data=) at mod_lua.cpp:476 L = mycmd = 0x7f43dc7536a0 "vm_check_msgs.lua" #19 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43dd344798, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412") at src/switch_core_session.c:2809 log = lp = event = 0x0 var = channel = 0x7f43dd34a380 expanded = 0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412" app = 0x7f43d245bc1d "lua" app_uuid_var = msg = {from = 0x7f43e4a3398e "src/switch_core_session.c", message_id = SWITCH_MESSAGE_INDICATE_APPLICATION_EXEC, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x7f43d245bc1d "lua", 0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412", 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} delim = scope = uuid_str = "ac9b4d90-7b71-4fed-84fa-99af0307bcad\000\000\000\000\357G\003\343C\177\000\000\n", '\000' "\200, \376q\002\000\000\000\000\034+\003\343C\177\000\000\060\000\000\000\000\000\000\000i\246\242\344C\177\000\000 \000\000\324C\177\000\000\n\000\000\000\000\000\000\000\220\221N\334C\177\000\000\000\000\000\000\000\000\000\000\260\030\062\324C\177\000\000\000,\003\343C\177\000\000\220\221N\334C\177\000\000i\246\242\344C\177\000\000\n\000\000\000\000\000\000\000a\347\213\344C\177\000\000\360j\351\001\000\000\000\000x\325\230\344C\177\000\000i\246\242\344C\177\000\000\230\023\311\001\000\000\000\000@\375\344\314C\177\000\000\004\000\000\000\000\000\000\000@\375\344\314C\177\000\000\000\000\000\000\000\000\000\000X\374\344\314C\177\000\000\262"... app_uuid = 0x7f43d0b0ba10 "ac9b4d90-7b71-4fed-84fa-99af0307bcad" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #20 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43dd344798, app=0x7f43cce4fd40 "lua", arg=0x7f43cce4fd48 "vm_check_msgs.lua default 212.225.254.22 952274412", flags=flags at entry=0x0) at src/switch_core_session.c:2684 application_interface = 0x7f43dc1c9d60 status = SWITCH_STATUS_SUCCESS __func__ = "switch_core_session_execute_application_get_flags" #21 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43dd344798) at src/switch_core_state_machine.c:230 current_application = extension = 0x7f43cce4fc58 uuid = #22 switch_core_session_run (session=0x7f43dd344798) at src/switch_core_state_machine.c:481 global_proceed = 1 index = proceed = do_extra_handlers = 1 ptr = rstatus = state = midstate = CS_EXECUTE endstate = endpoint_interface = driver_state_handler = 0x7f43dbdef600 application_state_handler = new_loops = 500 __PRETTY_FUNCTION__ = "switch_core_session_run" __func__ = "switch_core_session_run" #23 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43dd344798) at src/switch_core_session.c:1542 session = 0x7f43dd344798 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #24 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43cce85d80, obj=) at src/switch_core_session.c:1634 td = 0x7f43dccad1f8 check_status = node = pool = 0x7f43cce85b28 pop = 0x7f43dccad1f8 check = 0 __func__ = "switch_core_session_thread_pool_worker" #25 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #26 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #27 0x0000000000000000 in ?? () No symbol table info available. Thread 11 (Thread 0x7f43e52ee700 (LWP 10493)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 609014} #2 0x00007f43e48d9905 in pool_thread (thread=, obj=) at src/switch_core_memory.c:565 len = 0 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. Thread 10 (Thread 0x7f43d09a0700 (LWP 13119)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e5350f90, data=0x7f43d099fe60) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e48e7048 in switch_core_session_thread_pool_worker (thread=0x7f43cd9170d0, obj=) at src/switch_core_session.c:1615 check_status = node = pool = 0x7f43cd916e78 pop = 0x0 check = 0 __func__ = "switch_core_session_thread_pool_worker" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 9 (Thread 0x7f43e1cc0700 (LWP 10498)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x7f43e1e05e70, data=0x7f43e1cbfe70) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e494ffb3 in log_thread (t=, obj=) at src/switch_log.c:294 pop = 0x0 node = 0x0 binding = __func__ = "log_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 8 (Thread 0x7f43d811b700 (LWP 10786)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495ae4c in timer_next (timer=0x7f43d8119a60) at src/switch_time.c:683 private_info = 0x7f43ccb46148 cond_index = delta = #2 0x00007f43e48d68e8 in switch_core_timer_next (timer=timer at entry=0x7f43d8119a60) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #3 0x00007f43d2874cbc in read_stream_thread (thread=, obj=0x7f43d405f748) at /usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c:230 is_open = use_fh = 0x7f43d8119b20 olen = 320 abuf = "N\030\247\026\251\023I\017\247\t\036\003\314\373\241\363\247\353\316\344\345\336\060\332\376????\322\361\320p??\320\065\321G\322^\324T????\335Y\340>\343\207\346#\352\271\355\247\360\250\362\203\363\026\363\006\362&\361\366\360\202\361c\362k\363\177\364X\365\f\366\324\366\300\367\022\371\376\372)\375\314\376\256\377%\000\226\000p\001\325\002\327\004\264\a\000\v\363\rp\020\263\022\305\024\311\026\000\031Q\033g\035J\037\032!\251\"\332#\266$\177%C&\207&C&\244%*$\227!\203\036_\033H\030\212\025\032\023\025\021#\020A\020\317\020+\021\321\020\213\017I\r\000\n\256\005\212\000\332\372X\364G\355\317\346_\341\327??\331\301\327\273\326\344??\325\r\326\372\326\376\327\061\331\366\332;\335N\337\350\340\062\342]\343\314\344\244\346\223\350:\352?\353\244\353\337\353E\354\063\355\027\357\332\361\027\365\213\370\344\373\366\376\323\001K\004U\006t\b\324\n\037\r\363\016\354\017\060\020,\020\021\020\062\020\347\020\037\022d\023\064\024]\024\035\024\322\023\305\023:\024\f\025\f\026<\027Y\030Z\031Y\032[\033\247\034\020\036>\037X c!\352!\202!T \273\036\370\034\032\033\366\030\373\026\335\025\205\025l\025+\025^\024\274\022=\020\252\f\343\a(\002\202\373)\364"... fname = 0x7f43d8119c50 "/usr/local/freeswitch/sounds/music/16000/ponce-preludio-in-e-major.wav" source = 0x7f43d405f748 fh = {file_interface = 0x1ebf798, flags = 2061, fd = 0x0, samples = 2339086, samplerate = 16000, native_rate = 16000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 7999784292, speed = 0, memory_pool = 0x7f43cc908a18, prebuf = 65536, interval = 0, private_info = 0x7f43cc908b48, handler = 0x0, pos = 7999784292, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 0, silence_hits = 0, offset_pos = 0, samples_in = 2162688, samples_out = 0, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x7f43cc2ceed0, pre_buffer_data = 0x7f43cd22c208 "\371\005\266\004\256\002u", pre_buffer_datalen = 65536, file = 0x7f43d2875c48 "/usr/local/src/freeswitch/src/mod/formats/mod_local_stream/mod_local_stream.c", func = 0x7f43d2875e80 "read_stream_thread", line = 213, file_path = 0x7f43cc908b00 "/usr/local/freeswitch/sounds/music/16000/ponce-preludio-in-e-major.wav", spool_path = 0x0, prefix = 0x0, max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} cp = file_buf = "ponce-preludio-in-e-major.wav\000-arabesca.wav\000wav", '\000' path_buf = "/usr/local/freeswitch/sounds/music/16000/ponce-preludio-in-e-major.wav\000-arabesca.wav\000wav", '\000' timer = {interval = 20, flags = 0, samples = 320, samplecount = 2116480, timer_interface = 0x1c93450, memory_pool = 0x7f43ccb45e18, private_info = 0x7f43ccb46148, diff = 0, tick = 0} fd = -1 audio_buffer = 0x1ec8d80 dist_buf = 0x1f2b5c8 "c\022\306\021\f\020\063\016\070\f\372\n\351\n\261\nI\tn\006\323\002:" used = skip = 0 temp_pool = 0x7f43ccb45e18 __func__ = "read_stream_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 7 (Thread 0x7f43e00c1700 (LWP 10507)): #0 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43dbb51e19 in su_epoll_port_wait_events (self=0x1c9e950, tout=) at su_epoll_port.c:495 j = n = events = 0 index = version = 1 ev = 0x7f43e00c0920 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00007f43dbb513f3 in su_base_port_step (self=0x1c9e950, tout=1000) at su_base_port.c:467 now = {tv_sec = 3624630166, tv_usec = 551721} __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00007f43dba73ddb in sofia_profile_thread_run (thread=, obj=0x7f43dc126950) at sofia.c:2664 profile = 0x7f43dc126950 node = 0x0 s_event = 0x0 use_100rel = use_timer = use_rfc_5626 = supported = 0x7f43dc128748 "timer, path, replaces" sanity = worker_thread = 0x7f43dba13b50 st = SWITCH_STATUS_SUCCESS qname = "sofia:internal", '\000' __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 6 (Thread 0x7f43e0c0e700 (LWP 10503)): #0 0x00007f43e308cbe3 in select () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43e4989c35 in apr_sleep (t=) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 669760} #2 0x00007f43e48e59cd in switch_core_sql_db_thread (thread=, obj=) at src/switch_core_sqldb.c:1238 sec = 8 reg_sec = 8 #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. Thread 5 (Thread 0x7f43d80a3700 (LWP 10788)): #0 0x00007f43e3e0b18d in read () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e495b822 in softtimer_runtime () at src/switch_time.c:940 exp = 1 too_late = 20000000 current_ms = 3480 x = tick = 44 sps_interval_ticks = ts = 1415641366609489 last = fwd_errs = rev_errs = 0 profile_tick = 0 tfd = 27 time_sync = 4 last_MICROSECONDS_PER_TICK = 20000 spec = {it_interval = {tv_sec = 0, tv_nsec = 20000000}, it_value = {tv_sec = 0, tv_nsec = 20000000}} __func__ = "softtimer_runtime" #2 0x00007f43e4900df3 in switch_loadable_module_exec (thread=0x1c92700, obj=0x1c921e0) at src/switch_loadable_module.c:98 status = ts = 0x1c921e0 module = 0x1c93560 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #3 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #4 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #5 0x0000000000000000 in ?? () No symbol table info available. Thread 4 (Thread 0x7f43e0084700 (LWP 10540)): #0 0x00007f43e3093743 in epoll_wait () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #1 0x00007f43dbb51e19 in su_epoll_port_wait_events (self=0x1ccfe80, tout=) at su_epoll_port.c:495 j = n = events = 0 index = version = 3 ev = 0x7f43e0083da0 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00007f43dbb5128d in su_base_port_run (self=0x1ccfe80) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00007f43dbb561a8 in su_pthread_port_clone_main (varg=0x7f43e00c07d0) at su_pthread_port.c:343 arg = 0x0 task = {{sut_port = 0x1ccfe80, sut_root = 0x1cd06f0}} zap = 1 #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 3 (Thread 0x7f43e10d5700 (LWP 10502)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e48e0e22 in switch_user_sql_thread (thread=, obj=0x7f43dc0fb170) at src/switch_core_sqldb.c:1894 i = lc = written = iterations = qm = 0x7f43dc0fb170 i = __func__ = "switch_user_sql_thread" #2 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #3 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #4 0x0000000000000000 in ?? () No symbol table info available. Thread 2 (Thread 0x7f43e1da3700 (LWP 10495)): #0 0x00007f43e3e082d4 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #1 0x00007f43e4978f51 in apr_queue_pop (queue=0x1b40880, data=0x7f43e1da2e50) at misc/apr_queue.c:276 rv = #2 0x00007f43e48bf885 in switch_queue_pop (queue=, data=) at src/switch_apr.c:1056 No locals. #3 0x00007f43e490d889 in switch_event_dispatch_thread (thread=, obj=0x1b40880) at src/switch_event.c:305 pop = 0x0 event = 0x0 queue = 0x1b40880 my_id = __func__ = "switch_event_dispatch_thread" #4 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #5 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #6 0x0000000000000000 in ?? () No symbol table info available. Thread 1 (Thread 0x7f43d0a91700 (LWP 12924)): #0 0x00007f43e497565a in CoreSession::recordFile (this=0x7f43dd74e860, file_name=0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", time_limit=30, silence_threshold=, silence_hits=) at src/switch_cpp.cpp:1151 status = SWITCH_STATUS_BREAK local_fh = {file_interface = 0x1ebf798, flags = 14, fd = 0x0, samples = 0, samplerate = 8000, native_rate = 8000, channels = 1, format = 65538, sections = 1, seekable = 1, sample_count = 169280, speed = 0, memory_pool = 0x0, prebuf = 0, interval = 0, private_info = 0x7f43cd5595d0, handler = 0x0, pos = 0, audio_buffer = 0x0, sp_audio_buffer = 0x0, thresh = 30, silence_hits = 250, offset_pos = 0, samples_in = 0, samples_out = 169280, vol = 0, resampler = 0x0, buffer = 0x0, dbuf = 0x0, dbuflen = 0, pre_buffer = 0x0, pre_buffer_data = 0x7f43cdd9b0e8 "\370\377\370\377\b", pre_buffer_datalen = 65536, file = 0x7f43e4a40bc5 "src/switch_ivr_play_say.c", func = 0x7f43e4a413d0 "switch_ivr_record_file", line = 518, file_path = 0x7f43cd559570 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav", spool_path = 0x0, prefix = 0x7f43ccf04fc0 "/usr/local/freeswitch/sounds", max_samples = 0, params = 0x0, cur_channels = 0, cur_samplerate = 0} __func__ = "recordFile" #1 0x00007f43d243b826 in _wrap_CoreSession_recordFile (L=0x7f43cc25d370) at mod_lua_wrap.cpp:4947 arg2 = 0x279bb78 "/usr/local/freeswitch/storage/voicemail/default/212.225.254.22/957436584/greeting_temp.wav" arg4 = 30 SWIG_arg = -1 arg1 = 0x7f43dd74e860 arg3 = 30 arg5 = result = #2 0x00007f43d2444f64 in luaD_precall (L=L at entry=0x7f43cc25d370, func=0x7f43cce39e80, nresults=nresults at entry=0) at ldo.c:319 ci = n = cl = funcr = #3 0x00007f43d244e348 in luaV_execute (L=L at entry=0x7f43cc25d370, nexeccalls=nexeccalls at entry=1) at lvm.c:587 b = nresults = 0 i = ra = cl = 0x7f43de975570 base = k = 0x7f43dcd02e20 pc = 0x7f43dcbdcd2c #4 0x00007f43d244538d in luaD_call (L=0x7f43cc25d370, func=0x7f43ccd1a6d0, nResults=) at ldo.c:377 No locals. #5 0x00007f43d244463a in luaD_rawrunprotected (L=L at entry=0x7f43cc25d370, f=f at entry=0x7f43d24417d0 , ud=ud at entry=0x7f43d0a90370) at ldo.c:116 lj = {previous = 0x0, b = {{__jmpbuf = {139929164567408, -7100896249400093559, 1, 139929165732864, 139929165732864, 0, 7202149886429140105, 7202145254174135433}, __mask_was_saved = 0, __saved_mask = {__val = {40, 14754083699, 80, 14653814306, 7234318537360108649, 139929164567592, 139929430523936, 40, 139929164567408, 0, 0, 139929548172288, 40, 139929164567592, 40, 0}}}}, status = 0} #6 0x00007f43d244553f in luaD_pcall (L=L at entry=0x7f43cc25d370, func=func at entry=0x7f43d24417d0 , u=u at entry=0x7f43d0a90370, old_top=, ef=) at ldo.c:463 status = oldnCcalls = 0 old_ci = 0 old_allowhooks = 1 '\001' old_errfunc = 0 #7 0x00007f43d2442be1 in lua_pcall (L=0x7f43cc25d370, nargs=0, nresults=0, errfunc=) at lapi.c:821 c = {func = 0x7f43ccd1a6d0, nresults = 0} status = func = #8 0x00007f43d24277d6 in docall (L=0x7f43cc25d370, narg=0, nresults=0, perror=0, fatal=1) at mod_lua.cpp:92 status = base = 1 #9 0x00007f43d2427e08 in lua_parse_and_execute (L=L at entry=0x7f43cc25d370, input_code=, input_code at entry=0x7f43dc513c10 "vm_check_msgs.lua") at mod_lua.cpp:195 file = fdup = 0x7f43dd15be60 "/usr/local/freeswitch/scripts/vm_check_msgs.lua" args = error = __func__ = "lua_parse_and_execute" __PRETTY_FUNCTION__ = "int lua_parse_and_execute(lua_State*, char*)" #10 0x00007f43d24288ed in lua_function (session=0x7f43ccd3c5b8, data=) at mod_lua.cpp:476 L = mycmd = 0x7f43dc513c10 "vm_check_msgs.lua" #11 0x00007f43e48ed18b in switch_core_session_exec (session=session at entry=0x7f43ccd3c5b8, application_interface=application_interface at entry=0x7f43dc1c9d60, arg=arg at entry=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584") at src/switch_core_session.c:2809 log = lp = event = 0x0 var = channel = 0x7f43ccd421a0 expanded = 0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584" app = 0x7f43d245bc1d "lua" app_uuid_var = msg = {from = 0x7f43e4a3398e "src/switch_core_session.c", message_id = SWITCH_MESSAGE_INDICATE_APPLICATION_EXEC, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x7f43d245bc1d "lua", 0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} delim = scope = uuid_str = "262a6b64-03cc-4bdf-b374-264e20e85380\000\000\000\000\300\275?\315C\177\000\000\n", '\000' "\340, \223L\314C\177\000\000\034+\003\343C\177\000\000\060\000\000\000\000\000\000\000i\246\242\344C\177\000\000`\336\063\343C\177\000\000\n\000\000\000\000\000\000\000p\023\017\314C\177\000\000\000\000\000\000\000\000\000\000??\223\003\000\000\000\000\000,\003\343C\177\000\000p\023\017\314C\177\000\000i\246\242\344C\177\000\000\n\000\000\000\000\000\000\000a\347\213\344C\177\000\000\360j\351\001\000\000\000\000x\325\230\344C\177\000\000i\246\242\344C\177\000\000\230\023\311\001\000\000\000\000\020\344\061\326C\177\000\000\004\000\000\000\000\000\000\000\020\344\061\326C\177\000\000\000\000\000\000\000\000\000\000(\343"... app_uuid = 0x7f43d0a90a10 "262a6b64-03cc-4bdf-b374-264e20e85380" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #12 0x00007f43e48ed63b in switch_core_session_execute_application_get_flags (session=session at entry=0x7f43ccd3c5b8, app=0x7f43d631e410 "lua", arg=0x7f43d631e418 "vm_check_msgs.lua default 212.225.254.22 957436584", flags=flags at entry=0x0) at src/switch_core_session.c:2684 application_interface = 0x7f43dc1c9d60 status = SWITCH_STATUS_SUCCESS __func__ = "switch_core_session_execute_application_get_flags" #13 0x00007f43e48f00c2 in switch_core_standard_on_execute (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:230 current_application = extension = 0x7f43d631e328 uuid = #14 switch_core_session_run (session=0x7f43ccd3c5b8) at src/switch_core_state_machine.c:481 global_proceed = 1 index = proceed = do_extra_handlers = 1 ptr = rstatus = state = midstate = CS_EXECUTE endstate = endpoint_interface = driver_state_handler = 0x7f43dbdef600 application_state_handler = new_loops = 500 __PRETTY_FUNCTION__ = "switch_core_session_run" __func__ = "switch_core_session_run" #15 0x00007f43e48ea83e in switch_core_session_thread (thread=, obj=0x7f43ccd3c5b8) at src/switch_core_session.c:1542 session = 0x7f43ccd3c5b8 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #16 0x00007f43e48e6f60 in switch_core_session_thread_pool_worker (thread=0x7f43c8c24000, obj=) at src/switch_core_session.c:1634 td = 0x7f43cc9077a8 check_status = node = pool = 0x7f43c8c23da8 pop = 0x7f43cc9077a8 check = 0 __func__ = "switch_core_session_thread_pool_worker" #17 0x00007f43e3e03b50 in start_thread () from /lib/x86_64-linux-gnu/libpthread.so.0 No symbol table info available. #18 0x00007f43e30930ed in clone () from /lib/x86_64-linux-gnu/libc.so.6 No symbol table info available. #19 0x0000000000000000 in ?? () No symbol table info available. From lists at telefaks.de Tue Nov 11 15:27:34 2014 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 11 Nov 2014 13:27:34 +0100 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address In-Reply-To: References: <545CB9C0.1030807@telefaks.de> Message-ID: <546200B6.2000706@telefaks.de> Hello Brian, a reloadacl shows 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1306 Created ip list rfc6598.auto default (deny) 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 100.64.0.0/10 (allow) [] to list rfc6598.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1312 Created ip list rfc1918.auto default (deny) 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 (allow) [] to list rfc1918.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 172.16.0.0/12 (allow) [] to list rfc1918.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 192.168.0.0/16 (allow) [] to list rfc1918.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1320 Created ip list wan.auto default (allow) 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 0.0.0.0/8 (deny) [] to list wan.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 (deny) [] to list wan.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 172.16.0.0/12 (deny) [] to list wan.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 192.168.0.0/16 (deny) [] to list wan.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 169.254.0.0/16 (deny) [] to list wan.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1330 Created ip list nat.auto default (deny) 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1332 Adding 192.168.206.241/255.255.255.255 (deny) to list nat.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 (allow) [] to list nat.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 172.16.0.0/12 (allow) [] to list nat.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 192.168.0.0/16 (allow) [] to list nat.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 100.64.0.0/10 (allow) [] to list nat.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1342 Created ip list loopback.auto default (deny) 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 127.0.0.0/8 (allow) [] to list loopback.auto 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1348 Created ip list localnet.auto default (deny) 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1351 Adding 192.168.206.241/255.255.255.255 (allow) to list localnet.auto 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1376 Created ip list lan default (allow) 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding 192.168.42.0/24 (deny) [] to list lan 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding 192.168.42.0/24 (deny) to list lan 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding 192.168.42.42/32 (allow) [] to list lan 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding 192.168.42.42/32 (allow) to list lan 212.xxx.xxx.106 is set for ext-rtp-ip and ext-sip-ip. Best regards Peter On 11/07/14 14:16, Brian West wrote: > what do you have local-network-acl, ext-rtp-ip and ext-sip-ip? > > On Fri, Nov 7, 2014 at 6:23 AM, Peter Steinbach > wrote: > > Hell, > > we have the following problem: > > Our Freswitch is behind NAT. We are sending faxes to a SIP provider. > Dependend on the destination number, the faxes are received or not. > Faxes are always sent via the same SIP provider and the same dialplan, > but I expect, they may be routed differently via other, subsequent SIP > providers. > > Regarding the SDP I can see, that the c-line does contain the our > external IP, but the o-Line does contain the local IP in the > unicast_address field. > We are routing the call via a defined gateway in the external profile, > which has external_sip_ip set and external_rtp_ip set. > Dialplan is: > > > data="sofia/gateway/QSC/06912345678 at sip.qsc.de > "/> > > Here is the SDP > ==================================== > v=0. > o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. > s=FreeSWITCH. > c=IN IP4 212.xxx.xxx.106. > t=0 0. > m=audio 12056 RTP/AVP 0 8. > a=rtpmap:0 PCMU/8000/1. > a=rtpmap:8 PCMA/8000/1. > a=maxptime:240. > ==================================== > > I suspect, that the following SIP providers may have a problem > with the > o-line with the local IP. > So - Is there any way to control this? E.g. via Dialplan variable? > > ACL also seems to be fine > 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding > 192.168.0.0/16 (allow) [] to list rfc1918.auto > 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > > I see the same behaviour also in > https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with > proxy_media turned on" > There is a link for a patch, which is no longer available. Did this go > into the main release? Does anybody have this patch? > > > -- > With kind regards > Peter > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > > */Brian West/* > brian at freeswitch.org > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/5f4ce5bd/attachment.html From brian at freeswitch.org Tue Nov 11 16:24:26 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 07:24:26 -0600 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address In-Reply-To: <546200B6.2000706@telefaks.de> References: <545CB9C0.1030807@telefaks.de> <546200B6.2000706@telefaks.de> Message-ID: What you probably wanna do is set your ext-sip-ip and ext-rtp-ip to "autonat:212.xxx.xxx.106", then set the local-network-acl to nat.auto, Then restart FreeSWITCH. But what you outline doesn't make sense without autonat being on it should always use the ext-sip-ip and ext-rtp-ip on all outbound sip requests... can you get a sip trace of the one you're saying is exhibiting this issue? On Tue, Nov 11, 2014 at 6:27 AM, Peter Steinbach wrote: > Hello Brian, > > a reloadacl shows > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1306 Created ip list > rfc6598.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list rfc6598.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1312 Created ip list > rfc1918.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list rfc1918.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1320 Created ip list > wan.auto default (allow) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 0.0.0.0/8 > (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 169.254.0.0/16 (deny) [] to list wan.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1330 Created ip list > nat.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1332 Adding > 192.168.206.241/255.255.255.255 (deny) to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list nat.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1342 Created ip list > loopback.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 127.0.0.0/8 > (allow) [] to list loopback.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1348 Created ip list > localnet.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1351 Adding > 192.168.206.241/255.255.255.255 (allow) to list localnet.auto > > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1376 Created ip list lan > default (allow) > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.0/24 (deny) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.0/24 (deny) to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.42/32 (allow) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.42/32 (allow) to list lan > > 212.xxx.xxx.106 is set for ext-rtp-ip and ext-sip-ip. > > Best regards > Peter > > > On 11/07/14 14:16, Brian West wrote: > > what do you have local-network-acl, ext-rtp-ip and ext-sip-ip? > > On Fri, Nov 7, 2014 at 6:23 AM, Peter Steinbach wrote: > >> Hell, >> >> we have the following problem: >> >> Our Freswitch is behind NAT. We are sending faxes to a SIP provider. >> Dependend on the destination number, the faxes are received or not. >> Faxes are always sent via the same SIP provider and the same dialplan, >> but I expect, they may be routed differently via other, subsequent SIP >> providers. >> >> Regarding the SDP I can see, that the c-line does contain the our >> external IP, but the o-Line does contain the local IP in the >> unicast_address field. >> We are routing the call via a defined gateway in the external profile, >> which has external_sip_ip set and external_rtp_ip set. >> Dialplan is: >> >> >> > data="sofia/gateway/QSC/06912345678 at sip.qsc.de"/> >> >> Here is the SDP >> ==================================== >> v=0. >> o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. >> s=FreeSWITCH. >> c=IN IP4 212.xxx.xxx.106. >> t=0 0. >> m=audio 12056 RTP/AVP 0 8. >> a=rtpmap:0 PCMU/8000/1. >> a=rtpmap:8 PCMA/8000/1. >> a=maxptime:240. >> ==================================== >> >> I suspect, that the following SIP providers may have a problem with the >> o-line with the local IP. >> So - Is there any way to control this? E.g. via Dialplan variable? >> >> ACL also seems to be fine >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list rfc1918.auto >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list nat.auto >> >> I see the same behaviour also in >> https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with >> proxy_media turned on" >> There is a link for a patch, which is no longer available. Did this go >> into the main release? Does anybody have this patch? >> >> >> -- >> With kind regards >> Peter >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/aa4bba73/attachment-0001.html From brian at freeswitch.org Tue Nov 11 16:25:41 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 07:25:41 -0600 Subject: [Freeswitch-users] Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit In-Reply-To: References: Message-ID: The mailing list is not the correct place to report bugs, JIRA is, sadly 1.2.x is EOL and no longer supported. Is this an issue you can replicate on demand? On Tue, Nov 11, 2014 at 5:33 AM, Eduardo Alonso wrote: > Dear list: > > This is my first mail to the list. > I hope that my data and email are well formed. > > We are using freeswitch as voicemail in a production environment and every > week we obtain a core dump like the attached file. > > The error occurs in the CoreSession::recordFile function when the customer > try to record a greeting message. > > As is normal it's not possible to attach the entire core file, as well I > attached the backtrace following the debugging guide in the wiki: > https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. > I hope that the attached file help you to find the error. If more > information or testing is needed please let me know. > I'm looking for a similar core in the list, but all errors related with > this issue are different, from my point of view. > > Thank you in advance for your support. > > Cheers and best regards. > -- > *Eduardo Alonso Gil* > VoIP Systems Engineer @ Quobis | e: > eduardo.alonso at quobis.com | t: +34902999465 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/b715327e/attachment.html From brian at freeswitch.org Tue Nov 11 16:29:31 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 07:29:31 -0600 Subject: [Freeswitch-users] UDP vs UDP-NAT In-Reply-To: <5461C7E9.5090209@openlogic.com.au> References: <5461C7E9.5090209@openlogic.com.au> Message-ID: Can you look at the registered contacts: at fs_cli: sofia status profile internal reg On Tue, Nov 11, 2014 at 2:25 AM, Dion Phillips wrote: > Hi All > > I have a FS server in the cloud and have phones registering with FS from 2 > different locations. From one location, the registration is UDP-NAT and > just UDP from the second location. > > In the FS log, calls to the UDP connected phones work and they come up > with an external IP address (sofia_glue.c:1232 > sofia/internal/sip:1002 at 203.59.xxx.xxx:5060 sending invite) while the > phones on the UDP-NAT site don't work because FS is trying to connect to > them via the internal IP address ( sofia_glue.c:1232 > sofia/internal/sip:1000 at 192.168.0.111:5060 sending invite). > > The phones on both sites are behind a very standard modem with no port > forwarding or NAT setting. > > What is wrong and how would I fix it? > > Thanks in advance. > Dion. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/cabf5b38/attachment.html From avi at avimarcus.net Tue Nov 11 16:31:13 2014 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 11 Nov 2014 13:31:13 +0000 Subject: [Freeswitch-users] what can I have in ${} construction In-Reply-To: References: <30097.1355410423@ccs.covici.com> Message-ID: <000001499f0d0fd8-7351ccf4-4547-463f-900e-6d2ebe2c565f-000000@email.amazonses.com> Hi - where on confluence is this useful information documented? When searching, I couldn't find even find it on the wiki. -Avi On Thu, Dec 13, 2012 at 7:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It supports the substring with positive and negative vals so: > > ${foo:3:4} would start at char 3 in the string and eval to the next 4 > chars > ${foo:-3:2} would start at the end and go back 3 chars then print the next > 2 chars > > If the var name is followed by a ( or a space, it will pass the values to > the FSAPI and expand the result inline. > > ${sofia_contact 1004} > > equiv of... > > ${sofia_contact(1004)} > > > > > > > > On Thu, Dec 13, 2012 at 8:53 AM, wrote: > >> Hi. I was experimenting with ${variable} and was wondering what could >> be in there before the }? I can do a substring if I say variable:2, but >> I wonder what else you can do? The bash manual has all kinds of things >> you can have, but how much of this will fs do? >> >> 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 >> covici at ccs.covici.com >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/5acf1ed4/attachment.html From brian at freeswitch.org Tue Nov 11 16:35:27 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 07:35:27 -0600 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: just record to a file without any extension. 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz Now you'll be all set to waste more time dealing with two files vs the single wav file. On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: > In this specific case, I'm using g711, but can have either a-law or u-law > but the service that requires the recording understand only a-law, at the > moment I'm using ffmpeg to convert after the call ended, and then send it > to that service. > > Thanks > Ido > On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" > wrote: > >> as far as I understand, FreeSWITCH chooses the WAV parameters which >> are matching the current channel sampling frequency and codec. For >> example, if you record a G722 call, you would get a 16kHZ WAV file. >> >> Forcing it into one specific encoding would add real-time CPU load. I >> think it's still more preferable to run a post-processing job in low >> priority -- this way you ensure that the ongoing calls get the best >> serving. >> >> You can easily catch the event of call ending and trigger the >> conversion job, by listening to the events via ESL connection. >> >> >> >> >> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >> > Thinking in learning better the source code, and might create my first >> patch >> > to FS :) >> > >> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du wrote: >> >> >> >> No. Maybe bounty it? >> >> >> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >> >> >> >> Hello, >> >> >> >> I could not find any documentation on this subject. >> >> I require to have a recording of an a-law wav file (RIFF >> (little-endian) >> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >> >> >> >> Can I set a variable that tells the record command what type of codec >> to >> >> store the file with ? >> >> >> >> I know how to convert a normal wav recording into it using ffmpeg (for >> >> example): >> >> >> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >> >> >> >> But I wish to avoid it, if I can use the record app, or use the >> >> uuid_record API for it. >> >> >> >> Thanks, >> >> Ido >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://confluence.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://confluence.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/ce2fc430/attachment-0001.html From ben at langfeld.co.uk Tue Nov 11 16:49:43 2014 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 11 Nov 2014 11:49:43 -0200 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: Brian, was this answer supposed to be to some other question? The question here was about transcoding the recording, not about stereo vs two mono recordings... On 11 November 2014 11:35, Brian West wrote: > just record to a file without any extension. > > > > > > > > > > > > > > > > > > > > > > > > > > > > 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File > [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz > > 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File > [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz > > Now you'll be all set to waste more time dealing with two files vs the > single wav file. > > On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: > >> In this specific case, I'm using g711, but can have either a-law or u-law >> but the service that requires the recording understand only a-law, at the >> moment I'm using ffmpeg to convert after the call ended, and then send it >> to that service. >> >> Thanks >> Ido >> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >> wrote: >> >>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>> are matching the current channel sampling frequency and codec. For >>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>> >>> Forcing it into one specific encoding would add real-time CPU load. I >>> think it's still more preferable to run a post-processing job in low >>> priority -- this way you ensure that the ongoing calls get the best >>> serving. >>> >>> You can easily catch the event of call ending and trigger the >>> conversion job, by listening to the events via ESL connection. >>> >>> >>> >>> >>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>> > Thinking in learning better the source code, and might create my first >>> patch >>> > to FS :) >>> > >>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du wrote: >>> >> >>> >> No. Maybe bounty it? >>> >> >>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>> >> >>> >> Hello, >>> >> >>> >> I could not find any documentation on this subject. >>> >> I require to have a recording of an a-law wav file (RIFF >>> (little-endian) >>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>> >> >>> >> Can I set a variable that tells the record command what type of codec >>> to >>> >> store the file with ? >>> >> >>> >> I know how to convert a normal wav recording into it using ffmpeg (for >>> >> example): >>> >> >>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>> >> >>> >> But I wish to avoid it, if I can use the record app, or use the >>> >> uuid_record API for it. >>> >> >>> >> Thanks, >>> >> Ido >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://confluence.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://confluence.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://confluence.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/1be7daf8/attachment.html From lists at telefaks.de Tue Nov 11 16:57:16 2014 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 11 Nov 2014 14:57:16 +0100 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address In-Reply-To: <546200B6.2000706@telefaks.de> References: <545CB9C0.1030807@telefaks.de> <546200B6.2000706@telefaks.de> Message-ID: <546215BC.2010301@telefaks.de> Hello Brian, I had the chance to have a time slot to test the behaviour with a brand new compiled Freeswitch from GIT. The Problem has disappeared. The previously tested Freeswitch was about 5 months old, so there must have been some fix in the meantime. Sorry for the inconvenience. Best regards Peter On 11/11/14 13:27, Peter Steinbach wrote: > Hello Brian, > > a reloadacl shows > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1306 Created ip list > rfc6598.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list rfc6598.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1312 Created ip list > rfc1918.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 10.0.0.0/8 (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list rfc1918.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1320 Created ip list > wan.auto default (allow) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 0.0.0.0/8 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 10.0.0.0/8 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 169.254.0.0/16 (deny) [] to list wan.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1330 Created ip list > nat.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1332 Adding > 192.168.206.241/255.255.255.255 (deny) to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 10.0.0.0/8 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list nat.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1342 Created ip list > loopback.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 127.0.0.0/8 (allow) [] to list loopback.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1348 Created ip list > localnet.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1351 Adding > 192.168.206.241/255.255.255.255 (allow) to list localnet.auto > > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1376 Created ip list > lan default (allow) > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.0/24 (deny) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.0/24 (deny) to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.42/32 (allow) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.42/32 (allow) to list lan > > 212.xxx.xxx.106 is set for ext-rtp-ip and ext-sip-ip. > > Best regards > Peter > > On 11/07/14 14:16, Brian West wrote: >> what do you have local-network-acl, ext-rtp-ip and ext-sip-ip? >> >> On Fri, Nov 7, 2014 at 6:23 AM, Peter Steinbach > > wrote: >> >> Hell, >> >> we have the following problem: >> >> Our Freswitch is behind NAT. We are sending faxes to a SIP provider. >> Dependend on the destination number, the faxes are received or not. >> Faxes are always sent via the same SIP provider and the same >> dialplan, >> but I expect, they may be routed differently via other, >> subsequent SIP >> providers. >> >> Regarding the SDP I can see, that the c-line does contain the our >> external IP, but the o-Line does contain the local IP in the >> unicast_address field. >> We are routing the call via a defined gateway in the external >> profile, >> which has external_sip_ip set and external_rtp_ip set. >> Dialplan is: >> >> >> > data="sofia/gateway/QSC/06912345678 at sip.qsc.de >> "/> >> >> Here is the SDP >> ==================================== >> v=0. >> o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. >> s=FreeSWITCH. >> c=IN IP4 212.xxx.xxx.106. >> t=0 0. >> m=audio 12056 RTP/AVP 0 8. >> a=rtpmap:0 PCMU/8000/1. >> a=rtpmap:8 PCMA/8000/1. >> a=maxptime:240. >> ==================================== >> >> I suspect, that the following SIP providers may have a problem >> with the >> o-line with the local IP. >> So - Is there any way to control this? E.g. via Dialplan variable? >> >> ACL also seems to be fine >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list >> rfc1918.auto >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list nat.auto >> >> I see the same behaviour also in >> https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with >> proxy_media turned on" >> There is a link for a patch, which is no longer available. Did >> this go >> into the main release? Does anybody have this patch? >> >> >> -- >> With kind regards >> Peter >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> >> */Brian West/* >> brian at freeswitch.org >> >> >> */Twitter: @FreeSWITCH , @briankwest/* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/c94ebdce/attachment-0001.html From italorossib at gmail.com Tue Nov 11 17:08:18 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 11 Nov 2014 11:08:18 -0300 Subject: [Freeswitch-users] what can I have in ${} construction In-Reply-To: <000001499f0d0fd8-7351ccf4-4547-463f-900e-6d2ebe2c565f-000000@email.amazonses.com> References: <30097.1355410423@ccs.covici.com> <000001499f0d0fd8-7351ccf4-4547-463f-900e-6d2ebe2c565f-000000@email.amazonses.com> Message-ID: It should be included in https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Variables , we are in the process of moving this page to the confluence. There'll be a sprint next Friday (more information later) to migrate pages from old wiki to confluence, this page will have special attention. On Tue, Nov 11, 2014 at 10:31 AM, Avi Marcus wrote: > Hi - where on confluence is this useful information documented? When > searching, I couldn't find even find it on the wiki. > -Avi > > On Thu, Dec 13, 2012 at 7:45 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> It supports the substring with positive and negative vals so: >> >> ${foo:3:4} would start at char 3 in the string and eval to the next 4 >> chars >> ${foo:-3:2} would start at the end and go back 3 chars then print the >> next 2 chars >> >> If the var name is followed by a ( or a space, it will pass the values to >> the FSAPI and expand the result inline. >> >> ${sofia_contact 1004} >> >> equiv of... >> >> ${sofia_contact(1004)} >> >> >> >> >> >> >> >> On Thu, Dec 13, 2012 at 8:53 AM, wrote: >> >>> Hi. I was experimenting with ${variable} and was wondering what could >>> be in there before the }? I can do a substring if I say variable:2, but >>> I wonder what else you can do? The bash manual has all kinds of things >>> you can have, but how much of this will fs do? >>> >>> 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 >>> covici at ccs.covici.com >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/d3f0f099/attachment.html From dion at openlogic.com.au Tue Nov 11 17:10:37 2014 From: dion at openlogic.com.au (Dion Phillips) Date: Tue, 11 Nov 2014 22:10:37 +0800 Subject: [Freeswitch-users] UDP vs UDP-NAT In-Reply-To: References: <5461C7E9.5090209@openlogic.com.au> Message-ID: <546218DD.9060502@openlogic.com.au> Thanks Brian 'Contact:' is as follows UDP-NAT phones "" UDP phones "" (public ip addresses removed for security) What does that mean as I am not familiar with the internals for FS? Thanks Dion. On 11/11/14 21:29, Brian West wrote: > Can you look at the registered contacts: > > at fs_cli: > > sofia status profile internal reg > > > > On Tue, Nov 11, 2014 at 2:25 AM, Dion Phillips > wrote: > > Hi All > > I have a FS server in the cloud and have phones registering with > FS from 2 different locations. From one location, the registration > is UDP-NAT and just UDP from the second location. > > In the FS log, calls to the UDP connected phones work and they > come up with an external IP address (sofia_glue.c:1232 > sofia/internal/sip:1002 at 203.59.xxx.xxx:5060 > sending > invite) while the phones on the UDP-NAT site don't work because FS > is trying to connect to them via the internal IP address > (sofia_glue.c:1232 sofia/internal/sip:1000 at 192.168.0.111:5060 > sending invite). > > The phones on both sites are behind a very standard modem with no > port forwarding or NAT setting. > > What is wrong and how would I fix it? > > Thanks in advance. > Dion. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > > */Brian West/* > brian at freeswitch.org > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/166c238c/attachment.html From brian at freeswitch.org Tue Nov 11 17:25:13 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 08:25:13 -0600 Subject: [Freeswitch-users] UDP vs UDP-NAT In-Reply-To: <546218DD.9060502@openlogic.com.au> References: <5461C7E9.5090209@openlogic.com.au> <546218DD.9060502@openlogic.com.au> Message-ID: This is why I said you MUST look at the signaling, That fs_path is where it will really send the packet on the UDP-NAT registered device. "sofia global siptrace on" at fs_cli Place a call, you have some nat/crazyness going on here and you may need to first make sure the client side stops using port 5060, multiple devices behind the sane NAT will cause issues when fighting for 5060. What is the network topology? On Tue, Nov 11, 2014 at 8:10 AM, Dion Phillips wrote: > Thanks Brian > > 'Contact:' is as follows > > UDP-NAT phones > "" > > > > UDP phones > "" > > (public ip addresses removed for security) > > What does that mean as I am not familiar with the internals for FS? > > Thanks > Dion. > > > > > On 11/11/14 21:29, Brian West wrote: > > Can you look at the registered contacts: > > at fs_cli: > > sofia status profile internal reg > > > > On Tue, Nov 11, 2014 at 2:25 AM, Dion Phillips > wrote: > >> Hi All >> >> I have a FS server in the cloud and have phones registering with FS from >> 2 different locations. From one location, the registration is UDP-NAT and >> just UDP from the second location. >> >> In the FS log, calls to the UDP connected phones work and they come up >> with an external IP address (sofia_glue.c:1232 >> sofia/internal/sip:1002 at 203.59.xxx.xxx:5060 sending invite) while the >> phones on the UDP-NAT site don't work because FS is trying to connect to >> them via the internal IP address ( sofia_glue.c:1232 >> sofia/internal/sip:1000 at 192.168.0.111:5060 sending invite). >> >> The phones on both sites are behind a very standard modem with no port >> forwarding or NAT setting. >> >> What is wrong and how would I fix it? >> >> Thanks in advance. >> Dion. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/da22272c/attachment-0001.html From brian at freeswitch.org Tue Nov 11 17:26:38 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 08:26:38 -0600 Subject: [Freeswitch-users] NAT: SDP with local IP in o-line unicast_address In-Reply-To: <546215BC.2010301@telefaks.de> References: <545CB9C0.1030807@telefaks.de> <546200B6.2000706@telefaks.de> <546215BC.2010301@telefaks.de> Message-ID: Peter, Yes that bug was about 5 months old, this is why you always test with master or the latest release prior to mailing the list or opening a JIRA. ;) I just assumed you were using something rather recent. On Tue, Nov 11, 2014 at 7:57 AM, Peter Steinbach wrote: > Hello Brian, > > I had the chance to have a time slot to test the behaviour with a brand > new compiled Freeswitch from GIT. The Problem has disappeared. The > previously tested Freeswitch was about 5 months old, so there must have > been some fix in the meantime. > Sorry for the inconvenience. > > Best regards > Peter > > > > > > On 11/11/14 13:27, Peter Steinbach wrote: > > Hello Brian, > > a reloadacl shows > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1306 Created ip list > rfc6598.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list rfc6598.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1312 Created ip list > rfc1918.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list rfc1918.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list rfc1918.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1320 Created ip list > wan.auto default (allow) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 0.0.0.0/8 > (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (deny) [] to list wan.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 169.254.0.0/16 (deny) [] to list wan.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1330 Created ip list > nat.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1332 Adding > 192.168.206.241/255.255.255.255 (deny) to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 10.0.0.0/8 > (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 172.16.0.0/12 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 192.168.0.0/16 (allow) [] to list nat.auto > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding > 100.64.0.0/10 (allow) [] to list nat.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1342 Created ip list > loopback.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_utils.c:325 Adding 127.0.0.0/8 > (allow) [] to list loopback.auto > > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1348 Created ip list > localnet.auto default (deny) > 2014-11-11 13:21:37.079554 [NOTICE] switch_core.c:1351 Adding > 192.168.206.241/255.255.255.255 (allow) to list localnet.auto > > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1376 Created ip list lan > default (allow) > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.0/24 (deny) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.0/24 (deny) to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_utils.c:325 Adding > 192.168.42.42/32 (allow) [] to list lan > 2014-11-11 13:21:37.179566 [NOTICE] switch_core.c:1451 Adding > 192.168.42.42/32 (allow) to list lan > > 212.xxx.xxx.106 is set for ext-rtp-ip and ext-sip-ip. > > Best regards > Peter > > On 11/07/14 14:16, Brian West wrote: > > what do you have local-network-acl, ext-rtp-ip and ext-sip-ip? > > On Fri, Nov 7, 2014 at 6:23 AM, Peter Steinbach wrote: > >> Hell, >> >> we have the following problem: >> >> Our Freswitch is behind NAT. We are sending faxes to a SIP provider. >> Dependend on the destination number, the faxes are received or not. >> Faxes are always sent via the same SIP provider and the same dialplan, >> but I expect, they may be routed differently via other, subsequent SIP >> providers. >> >> Regarding the SDP I can see, that the c-line does contain the our >> external IP, but the o-Line does contain the local IP in the >> unicast_address field. >> We are routing the call via a defined gateway in the external profile, >> which has external_sip_ip set and external_rtp_ip set. >> Dialplan is: >> >> >> > data="sofia/gateway/QSC/06912345678 at sip.qsc.de"/> >> >> Here is the SDP >> ==================================== >> v=0. >> o=FreeSWITCH 1037989557 1037989558 IN IP4 192.168.206.241. >> s=FreeSWITCH. >> c=IN IP4 212.xxx.xxx.106. >> t=0 0. >> m=audio 12056 RTP/AVP 0 8. >> a=rtpmap:0 PCMU/8000/1. >> a=rtpmap:8 PCMA/8000/1. >> a=maxptime:240. >> ==================================== >> >> I suspect, that the following SIP providers may have a problem with the >> o-line with the local IP. >> So - Is there any way to control this? E.g. via Dialplan variable? >> >> ACL also seems to be fine >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list rfc1918.auto >> 2014-11-07 12:31:10.599563 [NOTICE] switch_utils.c:324 Adding >> 192.168.0.0/16 (allow) [] to list nat.auto >> >> I see the same behaviour also in >> https://freeswitch.org/jira/browse/FS-5909 "ext-xxx-ip ignored with >> proxy_media turned on" >> There is a link for a patch, which is no longer available. Did this go >> into the main release? Does anybody have this patch? >> >> >> -- >> With kind regards >> Peter >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/a03754ff/attachment.html From brian at freeswitch.org Tue Nov 11 17:33:30 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 08:33:30 -0600 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: If he wants to record without transcoding it is the only viable way to do so currently via mod_native_file. The task was a nice exercise as it exposed a nice little segfault in the stop record. (FS-6980). I fully understood what he wanted, and that it wasn't possible, offering up what we currently can do with mod_native_file. On Tue, Nov 11, 2014 at 7:49 AM, Ben Langfeld wrote: > Brian, was this answer supposed to be to some other question? The question > here was about transcoding the recording, not about stereo vs two mono > recordings... > > On 11 November 2014 11:35, Brian West wrote: > >> just record to a file without any extension. >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz >> >> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz >> >> Now you'll be all set to waste more time dealing with two files vs the >> single wav file. >> >> On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: >> >>> In this specific case, I'm using g711, but can have either a-law or >>> u-law but the service that requires the recording understand only a-law, at >>> the moment I'm using ffmpeg to convert after the call ended, and then send >>> it to that service. >>> >>> Thanks >>> Ido >>> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >>> wrote: >>> >>>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>>> are matching the current channel sampling frequency and codec. For >>>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>>> >>>> Forcing it into one specific encoding would add real-time CPU load. I >>>> think it's still more preferable to run a post-processing job in low >>>> priority -- this way you ensure that the ongoing calls get the best >>>> serving. >>>> >>>> You can easily catch the event of call ending and trigger the >>>> conversion job, by listening to the events via ESL connection. >>>> >>>> >>>> >>>> >>>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>>> > Thinking in learning better the source code, and might create my >>>> first patch >>>> > to FS :) >>>> > >>>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du >>>> wrote: >>>> >> >>>> >> No. Maybe bounty it? >>>> >> >>>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>>> >> >>>> >> Hello, >>>> >> >>>> >> I could not find any documentation on this subject. >>>> >> I require to have a recording of an a-law wav file (RIFF >>>> (little-endian) >>>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>>> >> >>>> >> Can I set a variable that tells the record command what type of >>>> codec to >>>> >> store the file with ? >>>> >> >>>> >> I know how to convert a normal wav recording into it using ffmpeg >>>> (for >>>> >> example): >>>> >> >>>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>>> >> >>>> >> But I wish to avoid it, if I can use the record app, or use the >>>> >> uuid_record API for it. >>>> >> >>>> >> Thanks, >>>> >> Ido >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://confluence.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>>> >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://confluence.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> > >>>> > >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://confluence.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/f68065bd/attachment-0001.html From brian at freeswitch.org Tue Nov 11 17:38:04 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 08:38:04 -0600 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: Looking at the libsndfile API this should be doable but I would need to read up on it more, If you require this feature email consulting at freeswitch.org for a quote on adding this functionality. Thanks On Tue, Nov 11, 2014 at 8:33 AM, Brian West wrote: > If he wants to record without transcoding it is the only viable way to do > so currently via mod_native_file. The task was a nice exercise as it > exposed a nice little segfault in the stop record. (FS-6980). > > I fully understood what he wanted, and that it wasn't possible, offering > up what we currently can do with mod_native_file. > > > On Tue, Nov 11, 2014 at 7:49 AM, Ben Langfeld wrote: > >> Brian, was this answer supposed to be to some other question? The >> question here was about transcoding the recording, not about stereo vs two >> mono recordings... >> >> On 11 November 2014 11:35, Brian West wrote: >> >>> just record to a file without any extension. >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz >>> >>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz >>> >>> Now you'll be all set to waste more time dealing with two files vs the >>> single wav file. >>> >>> On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: >>> >>>> In this specific case, I'm using g711, but can have either a-law or >>>> u-law but the service that requires the recording understand only a-law, at >>>> the moment I'm using ffmpeg to convert after the call ended, and then send >>>> it to that service. >>>> >>>> Thanks >>>> Ido >>>> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >>>> wrote: >>>> >>>>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>>>> are matching the current channel sampling frequency and codec. For >>>>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>>>> >>>>> Forcing it into one specific encoding would add real-time CPU load. I >>>>> think it's still more preferable to run a post-processing job in low >>>>> priority -- this way you ensure that the ongoing calls get the best >>>>> serving. >>>>> >>>>> You can easily catch the event of call ending and trigger the >>>>> conversion job, by listening to the events via ESL connection. >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>>>> > Thinking in learning better the source code, and might create my >>>>> first patch >>>>> > to FS :) >>>>> > >>>>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du >>>>> wrote: >>>>> >> >>>>> >> No. Maybe bounty it? >>>>> >> >>>>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>>>> >> >>>>> >> Hello, >>>>> >> >>>>> >> I could not find any documentation on this subject. >>>>> >> I require to have a recording of an a-law wav file (RIFF >>>>> (little-endian) >>>>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>>>> >> >>>>> >> Can I set a variable that tells the record command what type of >>>>> codec to >>>>> >> store the file with ? >>>>> >> >>>>> >> I know how to convert a normal wav recording into it using ffmpeg >>>>> (for >>>>> >> example): >>>>> >> >>>>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>>>> >> >>>>> >> But I wish to avoid it, if I can use the record app, or use the >>>>> >> uuid_record API for it. >>>>> >> >>>>> >> Thanks, >>>>> >> Ido >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.com >>>>> >> >>>>> >> Official FreeSWITCH Sites >>>>> >> http://www.freeswitch.org >>>>> >> http://confluence.freeswitch.org >>>>> >> http://www.cluecon.com >>>>> >> >>>>> >> FreeSWITCH-users mailing list >>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> >> >>>>> >> >>>>> >> >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.com >>>>> >> >>>>> >> Official FreeSWITCH Sites >>>>> >> http://www.freeswitch.org >>>>> >> http://confluence.freeswitch.org >>>>> >> http://www.cluecon.com >>>>> >> >>>>> >> FreeSWITCH-users mailing list >>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >> http://www.freeswitch.org >>>>> > >>>>> > >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://confluence.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/6c278684/attachment-0001.html From aqsyounas at gmail.com Tue Nov 11 17:38:40 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 11 Nov 2014 19:38:40 +0500 Subject: [Freeswitch-users] Number of ways for creating a dynamic ivr in freeswitch Message-ID: Hi, How can we create dynamic ivr in freeswitch. I have ivr data in json format from a web, i want this to be played in freeswitch as ivr. How can i do so.? In asterisk there are application availables like phpivr/SynIVR. How can this be done in Freeswitch. Any help would be much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/24b0ccb3/attachment.html From cmrienzo at gmail.com Tue Nov 11 17:46:30 2014 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 11 Nov 2014 09:46:30 -0500 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: Getting the lib to do it is easy. It's mainly an API problem- how will FS expose something like this? On Tue, Nov 11, 2014 at 9:38 AM, Brian West wrote: > Looking at the libsndfile API this should be doable but I would need to > read up on it more, If you require this feature email > consulting at freeswitch.org for a quote on adding this functionality. > > Thanks > > On Tue, Nov 11, 2014 at 8:33 AM, Brian West wrote: > >> If he wants to record without transcoding it is the only viable way to do >> so currently via mod_native_file. The task was a nice exercise as it >> exposed a nice little segfault in the stop record. (FS-6980). >> >> I fully understood what he wanted, and that it wasn't possible, offering >> up what we currently can do with mod_native_file. >> >> >> On Tue, Nov 11, 2014 at 7:49 AM, Ben Langfeld wrote: >> >>> Brian, was this answer supposed to be to some other question? The >>> question here was about transcoding the recording, not about stereo vs two >>> mono recordings... >>> >>> On 11 November 2014 11:35, Brian West wrote: >>> >>>> just record to a file without any extension. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> "/tmp/${uuid}-testing"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz >>>> >>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz >>>> >>>> Now you'll be all set to waste more time dealing with two files vs the >>>> single wav file. >>>> >>>> On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: >>>> >>>>> In this specific case, I'm using g711, but can have either a-law or >>>>> u-law but the service that requires the recording understand only a-law, at >>>>> the moment I'm using ffmpeg to convert after the call ended, and then send >>>>> it to that service. >>>>> >>>>> Thanks >>>>> Ido >>>>> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >>>>> wrote: >>>>> >>>>>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>>>>> are matching the current channel sampling frequency and codec. For >>>>>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>>>>> >>>>>> Forcing it into one specific encoding would add real-time CPU load. I >>>>>> think it's still more preferable to run a post-processing job in low >>>>>> priority -- this way you ensure that the ongoing calls get the best >>>>>> serving. >>>>>> >>>>>> You can easily catch the event of call ending and trigger the >>>>>> conversion job, by listening to the events via ESL connection. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>>>>> > Thinking in learning better the source code, and might create my >>>>>> first patch >>>>>> > to FS :) >>>>>> > >>>>>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du >>>>>> wrote: >>>>>> >> >>>>>> >> No. Maybe bounty it? >>>>>> >> >>>>>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>>>>> >> >>>>>> >> Hello, >>>>>> >> >>>>>> >> I could not find any documentation on this subject. >>>>>> >> I require to have a recording of an a-law wav file (RIFF >>>>>> (little-endian) >>>>>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>>>>> >> >>>>>> >> Can I set a variable that tells the record command what type of >>>>>> codec to >>>>>> >> store the file with ? >>>>>> >> >>>>>> >> I know how to convert a normal wav recording into it using ffmpeg >>>>>> (for >>>>>> >> example): >>>>>> >> >>>>>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>>>>> >> >>>>>> >> But I wish to avoid it, if I can use the record app, or use the >>>>>> >> uuid_record API for it. >>>>>> >> >>>>>> >> Thanks, >>>>>> >> Ido >>>>>> >> >>>>>> _________________________________________________________________________ >>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>> >> consulting at freeswitch.org >>>>>> >> http://www.freeswitchsolutions.com >>>>>> >> >>>>>> >> Official FreeSWITCH Sites >>>>>> >> http://www.freeswitch.org >>>>>> >> http://confluence.freeswitch.org >>>>>> >> http://www.cluecon.com >>>>>> >> >>>>>> >> FreeSWITCH-users mailing list >>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> _________________________________________________________________________ >>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>> >> consulting at freeswitch.org >>>>>> >> http://www.freeswitchsolutions.com >>>>>> >> >>>>>> >> Official FreeSWITCH Sites >>>>>> >> http://www.freeswitch.org >>>>>> >> http://confluence.freeswitch.org >>>>>> >> http://www.cluecon.com >>>>>> >> >>>>>> >> FreeSWITCH-users mailing list >>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> _________________________________________________________________________ >>>>>> > Professional FreeSWITCH Consulting Services: >>>>>> > consulting at freeswitch.org >>>>>> > http://www.freeswitchsolutions.com >>>>>> > >>>>>> > Official FreeSWITCH Sites >>>>>> > http://www.freeswitch.org >>>>>> > http://confluence.freeswitch.org >>>>>> > http://www.cluecon.com >>>>>> > >>>>>> > FreeSWITCH-users mailing list >>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> > UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> > http://www.freeswitch.org >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/56a3782b/attachment-0001.html From brian at freeswitch.org Tue Nov 11 17:59:58 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 08:59:58 -0600 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: I'm going to talk to Anthony about it shortly. On Tue, Nov 11, 2014 at 8:46 AM, Christopher Rienzo wrote: > Getting the lib to do it is easy. It's mainly an API problem- how will FS > expose something like this? > > On Tue, Nov 11, 2014 at 9:38 AM, Brian West wrote: > >> Looking at the libsndfile API this should be doable but I would need to >> read up on it more, If you require this feature email >> consulting at freeswitch.org for a quote on adding this functionality. >> >> Thanks >> >> On Tue, Nov 11, 2014 at 8:33 AM, Brian West wrote: >> >>> If he wants to record without transcoding it is the only viable way to >>> do so currently via mod_native_file. The task was a nice exercise as it >>> exposed a nice little segfault in the stop record. (FS-6980). >>> >>> I fully understood what he wanted, and that it wasn't possible, offering >>> up what we currently can do with mod_native_file. >>> >>> >>> On Tue, Nov 11, 2014 at 7:49 AM, Ben Langfeld >>> wrote: >>> >>>> Brian, was this answer supposed to be to some other question? The >>>> question here was about transcoding the recording, not about stereo vs two >>>> mono recordings... >>>> >>>> On 11 November 2014 11:35, Brian West wrote: >>>> >>>>> just record to a file without any extension. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> "/tmp/${uuid}-testing"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz >>>>> >>>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz >>>>> >>>>> Now you'll be all set to waste more time dealing with two files vs the >>>>> single wav file. >>>>> >>>>> On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: >>>>> >>>>>> In this specific case, I'm using g711, but can have either a-law or >>>>>> u-law but the service that requires the recording understand only a-law, at >>>>>> the moment I'm using ffmpeg to convert after the call ended, and then send >>>>>> it to that service. >>>>>> >>>>>> Thanks >>>>>> Ido >>>>>> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >>>>>> wrote: >>>>>> >>>>>>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>>>>>> are matching the current channel sampling frequency and codec. For >>>>>>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>>>>>> >>>>>>> Forcing it into one specific encoding would add real-time CPU load. I >>>>>>> think it's still more preferable to run a post-processing job in low >>>>>>> priority -- this way you ensure that the ongoing calls get the best >>>>>>> serving. >>>>>>> >>>>>>> You can easily catch the event of call ending and trigger the >>>>>>> conversion job, by listening to the events via ESL connection. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>>>>>> > Thinking in learning better the source code, and might create my >>>>>>> first patch >>>>>>> > to FS :) >>>>>>> > >>>>>>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du >>>>>>> wrote: >>>>>>> >> >>>>>>> >> No. Maybe bounty it? >>>>>>> >> >>>>>>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>>>>>> >> >>>>>>> >> Hello, >>>>>>> >> >>>>>>> >> I could not find any documentation on this subject. >>>>>>> >> I require to have a recording of an a-law wav file (RIFF >>>>>>> (little-endian) >>>>>>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>>>>>> >> >>>>>>> >> Can I set a variable that tells the record command what type of >>>>>>> codec to >>>>>>> >> store the file with ? >>>>>>> >> >>>>>>> >> I know how to convert a normal wav recording into it using ffmpeg >>>>>>> (for >>>>>>> >> example): >>>>>>> >> >>>>>>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>>>>>> >> >>>>>>> >> But I wish to avoid it, if I can use the record app, or use the >>>>>>> >> uuid_record API for it. >>>>>>> >> >>>>>>> >> Thanks, >>>>>>> >> Ido >>>>>>> >> >>>>>>> _________________________________________________________________________ >>>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>>> >> consulting at freeswitch.org >>>>>>> >> http://www.freeswitchsolutions.com >>>>>>> >> >>>>>>> >> Official FreeSWITCH Sites >>>>>>> >> http://www.freeswitch.org >>>>>>> >> http://confluence.freeswitch.org >>>>>>> >> http://www.cluecon.com >>>>>>> >> >>>>>>> >> FreeSWITCH-users mailing list >>>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >> http://www.freeswitch.org >>>>>>> >> >>>>>>> >> >>>>>>> >> >>>>>>> >> >>>>>>> _________________________________________________________________________ >>>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>>> >> consulting at freeswitch.org >>>>>>> >> http://www.freeswitchsolutions.com >>>>>>> >> >>>>>>> >> Official FreeSWITCH Sites >>>>>>> >> http://www.freeswitch.org >>>>>>> >> http://confluence.freeswitch.org >>>>>>> >> http://www.cluecon.com >>>>>>> >> >>>>>>> >> FreeSWITCH-users mailing list >>>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >> http://www.freeswitch.org >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> _________________________________________________________________________ >>>>>>> > Professional FreeSWITCH Consulting Services: >>>>>>> > consulting at freeswitch.org >>>>>>> > http://www.freeswitchsolutions.com >>>>>>> > >>>>>>> > Official FreeSWITCH Sites >>>>>>> > http://www.freeswitch.org >>>>>>> > http://confluence.freeswitch.org >>>>>>> > http://www.cluecon.com >>>>>>> > >>>>>>> > FreeSWITCH-users mailing list >>>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> > UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> > http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/b4f25180/attachment-0001.html From mbodbg at gmx.net Tue Nov 11 18:06:12 2014 From: mbodbg at gmx.net (mbo) Date: Tue, 11 Nov 2014 16:06:12 +0100 Subject: [Freeswitch-users] Freeswitch webRTC - audio DTLS key err In-Reply-To: References: <92D31E8B-D0B3-469C-B4EC-F0160C96B165@gmx.net> Message-ID: <91EA8EDA-F1EC-4C58-BA0B-08FE2F3D7B06@gmx.net> I?ve deleted dtls-srtp.pem (and dtls-srtp.key) and restartet freeswitch.The file dtls-srtp.pem was regenerated and the error does not appear anymore :-). One other question, the certificate dtls-srtp.pem was generated, but not the private key file dtls-srtp.key. Where can I find it? Thanks Markus Am 10.11.2014 um 18:32 schrieb Anthony Minessale : > You could run tshark on a terminal on the box and filter for dtls traffic to get a better idea. > Contrary to your statement, the dtls-srtp.pem is only relevant to WebRTC. You can try deleting that file and let FS generate a new one by restarting it. > > Also check the date and time on your box to make sure its correct. > > > > On Mon, Nov 10, 2014 at 10:53 AM, mbo wrote: > Hello, > > according to the instructions on https://freeswitch.org/confluence/display/FREESWITCH/WebRTC I have enabled webrtc on one Sip profile. I?ve added my own cerificate and installed in /usr/local/freeswitch/certs. I?m connecting to freeswitch using JsSip library. The signaling part works fine, I can see all SIP messages in the log, but then it fails to establish the audio/rtp connection: > > 2014-11-10 17:36:27.065688 [INFO] switch_core_media.c:5206 Skipping RTCP ICE (Same as RTP) > 2014-11-10 17:36:27.065688 [INFO] switch_rtp.c:3065 Activate RTP/RTCP audio DTLS client > 2014-11-10 17:36:27.065688 [ERR] switch_rtp.c:3117 audio DTLS key err [1] > > > I?m a bit confused about this message, I thought all tls settings in a sip profile and the dtls-srtp.pem certificate is not relevant for webrtc / wss, or what does this error message want to tell me? > > Thanks > > Markus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? http://freeswitch.org/g+ > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/e0a1306c/attachment.html From eduardo.alonso at quobis.com Tue Nov 11 18:30:12 2014 From: eduardo.alonso at quobis.com (Eduardo Alonso) Date: Tue, 11 Nov 2014 16:30:12 +0100 Subject: [Freeswitch-users] Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit In-Reply-To: References: Message-ID: Hello Brian: Thanks for you quick answer. I downgraded from 1.4 to 1.2 because the error occurred. I'm to open a ticket in JIRA with a core-dump file from 1.4. Thanks for your support. BR. 2014-11-11 14:25 GMT+01:00 Brian West : > The mailing list is not the correct place to report bugs, JIRA is, sadly > 1.2.x is EOL and no longer supported. Is this an issue you can replicate > on demand? > > On Tue, Nov 11, 2014 at 5:33 AM, Eduardo Alonso > wrote: > >> Dear list: >> >> This is my first mail to the list. >> I hope that my data and email are well formed. >> >> We are using freeswitch as voicemail in a production environment and >> every week we obtain a core dump like the attached file. >> >> The error occurs in the CoreSession::recordFile function when the >> customer try to record a greeting message. >> >> As is normal it's not possible to attach the entire core file, as well I >> attached the backtrace following the debugging guide in the wiki: >> https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. >> I hope that the attached file help you to find the error. If more >> information or testing is needed please let me know. >> I'm looking for a similar core in the list, but all errors related with >> this issue are different, from my point of view. >> >> Thank you in advance for your support. >> >> Cheers and best regards. >> -- >> *Eduardo Alonso Gil* >> VoIP Systems Engineer @ Quobis | e: >> eduardo.alonso at quobis.com | t: +34902999465 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis | e: eduardo.alonso at quobis.com | t: +34902999465 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/1fc117ac/attachment.html From dgarcia at anew.com.ve Tue Nov 11 19:02:35 2014 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 11 Nov 2014 11:32:35 -0430 Subject: [Freeswitch-users] Number of ways for creating a dynamic ivr in freeswitch In-Reply-To: References: Message-ID: <5462331B.5040505@anew.com.ve> Hi, Younas How are you doing? I think you get a reply before in your previous question. You have several ways of do it. Let start for one of them. FS has their api accesible from many lenguages: lua, perl, php, javascript, etc. Just, select one of them and play hard with it. I have and small IVR with lua for testing, lab and proof of concept and works nicely. You can use lua, get the json data from the web and control FS accordingly your needs. The script will be located inside of FS in a directory. Another way is using ESL. ESL is a tcp conectionto FS where you can control FS actions. As in the previous paragraph, you have to select a solution: a build one done in a lenguage (lua, php, ruby, javascript, etc) or build your own middleware/framework. The pros in this architecture is you can seperate ivr from FS and control several instances of FS, etc, etc, etc. One interesting example using FS ESL and node.js is: https://github.com/englercj/node-esl. I saw a project similar to your requirementet, instead of json using xml and spring framework but I lost the link. Other, references: http://docs.plivo.org/get-started/ http://m4d.colfinder.org/sites/default/files/Slides/M4D_Week5_setup_web_telephony_server.pdf I suggest you explore FS documentation, very good point to start, explore FS scripting and ESL. From there you will make up your own solution. On 11/11/2014 10:08 AM, Aqs Younas wrote: > Hi, > How can we create dynamic ivr in freeswitch. I have ivr data in json > format from a web, i want this to be played in freeswitch as ivr. How > can i do so.? > > In asterisk there are application availables like phpivr/SynIVR. > How can this be done in Freeswitch. > > Any help would be much appreciated. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/bb328c92/attachment-0001.html From aqsyounas at gmail.com Tue Nov 11 19:19:55 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Tue, 11 Nov 2014 21:19:55 +0500 Subject: [Freeswitch-users] Number of ways for creating a dynamic ivr in freeswitch In-Reply-To: <5462331B.5040505@anew.com.ve> References: <5462331B.5040505@anew.com.ve> Message-ID: Thanks. I do consider them. On 11 November 2014 21:02, Saugort Dario Garcia Tovar wrote: > Hi, Younas > > How are you doing? > > I think you get a reply before in your previous question. > > You have several ways of do it. > > Let start for one of them. > > FS has their api accesible from many lenguages: lua, perl, php, > javascript, etc. Just, select one of them and play hard with it. I have and > small IVR with lua for testing, lab and proof of concept and works nicely. > You can use lua, get the json data from the web and control FS accordingly > your needs. The script will be located inside of FS in a directory. > > Another way is using ESL. ESL is a tcp conectionto FS where you can > control FS actions. As in the previous paragraph, you have to select a > solution: a build one done in a lenguage (lua, php, ruby, javascript, etc) > or build your own middleware/framework. The pros in this architecture is > you can seperate ivr from FS and control several instances of FS, etc, etc, > etc. One interesting example using FS ESL and node.js is: > https://github.com/englercj/node-esl. > > I saw a project similar to your requirementet, instead of json using xml > and spring framework but I lost the link. Other, references: > http://docs.plivo.org/get-started/ > > http://m4d.colfinder.org/sites/default/files/Slides/M4D_Week5_setup_web_telephony_server.pdf > > I suggest you explore FS documentation, very good point to start, explore > FS scripting and ESL. From there you will make up your own solution. > > > > On 11/11/2014 10:08 AM, Aqs Younas wrote: > > Hi, > How can we create dynamic ivr in freeswitch. I have ivr data in json > format from a web, i want this to be played in freeswitch as ivr. How can i > do so.? > > In asterisk there are application availables like phpivr/SynIVR. > How can this be done in Freeswitch. > > Any help would be much appreciated. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/3c7ee857/attachment.html From krice at freeswitch.org Tue Nov 11 19:31:50 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 11 Nov 2014 10:31:50 -0600 Subject: [Freeswitch-users] Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit In-Reply-To: Message-ID: Please see https://freeswitch.org/confluence/display/FREESWITCH/Debugging for getting a backtrace. Opening a Jira and attach the backtrace. Please do not attach the core file. On 11/11/14 9:30 AM, "Eduardo Alonso" wrote: > Hello Brian: > > Thanks for you quick answer. > I downgraded from 1.4 to 1.2 because the error occurred. > > I'm to open a ticket in JIRA with a core-dump file from 1.4. > > Thanks for your support. > BR. > > 2014-11-11 14:25 GMT+01:00 Brian West : >> The mailing list is not the correct place to report bugs, JIRA is, sadly >> 1.2.x is EOL and no longer supported.? Is this an issue you can replicate on >> demand? >> >> On Tue, Nov 11, 2014 at 5:33 AM, Eduardo Alonso >> wrote: >>> Dear list: >>> >>> This is my first mail to the list. >>> I hope that my data and email are well formed. >>> >>> We are using freeswitch as voicemail in a production environment and every >>> week we obtain a core dump like the attached file. >>> >>> The error occurs in the CoreSession::recordFile function when the customer >>> try to record a greeting message. >>> >>> As is normal it's not possible to attach the entire core file, as well I >>> attached the backtrace following the debugging guide in the wiki: >>> https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. >>> I hope that the attached file help you to find the error. If more >>> information or testing is needed please let me know. >>> I'm looking for a similar core in the list, but all errors related with this >>> issue are different, from my point of view. >>> >>> Thank you in advance for your support. >>> >>> Cheers and best regards. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/1b272374/attachment.html From rossbcan at gmail.com Tue Nov 11 20:11:05 2014 From: rossbcan at gmail.com (Bill Ross) Date: Tue, 11 Nov 2014 12:11:05 -0500 Subject: [Freeswitch-users] RTP Triple NAT Traversal Message-ID: <000c01cffdd2$7b21b5c0$71652140$@gmail.com> Folks FS: 1.4.13 / OpenWrt Topology SIP Phone <-> WIFI AP / NAT <-> Cloud <-> Router / NAT <-> Router / NAT <-> FS 192.168.10.x Public IP 192.168.1.x 10.0.0.128 The SIP packets traverse the NAT chain correctly. The RTP packets from / to Phone are from / to 192.168.10.x, so, not getting through (because private N/W addresses embedded in SIP messaging) I am considering the following methods of handling this, but, before doing a lot of work / research, is there a better method? 1 - Using firewall rules (somehow, if possible, connection tracking?) to SNAT/DNAT packets to/from private network ranges which are not local lan range 2 - (appears simplest) Using channel variables (addresses) from SIP messaging to alter addresses prior to bridge. Any thoughts? Thanks; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/46fd9a6c/attachment.html From rtreleaven at bunnykick.ca Tue Nov 11 20:17:02 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Tue, 11 Nov 2014 12:17:02 -0500 Subject: [Freeswitch-users] RTP Triple NAT Traversal In-Reply-To: <000c01cffdd2$7b21b5c0$71652140$@gmail.com> References: <000c01cffdd2$7b21b5c0$71652140$@gmail.com> Message-ID: how bout a tunnel? On Tue, Nov 11, 2014 at 12:11 PM, Bill Ross wrote: > Folks > > > > FS: 1.4.13 / OpenWrt > > > > Topology > > > > SIP Phone <-> WIFI AP / NAT <-> Cloud <-> Router / > NAT <-> Router / NAT <-> FS > > > > 192.168.10.x Public > IP 192.168.1.x 10.0.0.128 > > > > The SIP packets traverse the NAT chain correctly. > > The RTP packets from / to Phone are from / to 192.168.10.x, so, not > getting through (because private N/W addresses embedded in SIP messaging) > > > > I am considering the following methods of handling this, but, before doing > a lot of work / research, is there a better method? > > > > 1 - Using firewall rules (somehow, if possible, connection tracking?) to > SNAT/DNAT packets to/from private network ranges which are not local lan > range > > 2 ? (appears simplest) Using channel variables (addresses) from SIP > messaging to alter addresses prior to bridge. > > > > Any thoughts? > > > > Thanks; > > Bill Ross > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/a21a7672/attachment-0001.html From brian at freeswitch.org Tue Nov 11 20:21:39 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 11 Nov 2014 11:21:39 -0600 Subject: [Freeswitch-users] RTP Triple NAT Traversal In-Reply-To: <000c01cffdd2$7b21b5c0$71652140$@gmail.com> References: <000c01cffdd2$7b21b5c0$71652140$@gmail.com> Message-ID: Your client should be doing the NAT Traversal in this case, using STUN to discover its public IP, then using that IP in the packets, along with enabling rport on the client side, that would help this greatly, FreeSWITCH will auto adjust the rest if you can get that media part to work. On Tue, Nov 11, 2014 at 11:11 AM, Bill Ross wrote: > Folks > > > > FS: 1.4.13 / OpenWrt > > > > Topology > > > > SIP Phone <-> WIFI AP / NAT <-> Cloud <-> Router / > NAT <-> Router / NAT <-> FS > > > > 192.168.10.x Public > IP 192.168.1.x 10.0.0.128 > > > > The SIP packets traverse the NAT chain correctly. > > The RTP packets from / to Phone are from / to 192.168.10.x, so, not > getting through (because private N/W addresses embedded in SIP messaging) > > > > I am considering the following methods of handling this, but, before doing > a lot of work / research, is there a better method? > > > > 1 - Using firewall rules (somehow, if possible, connection tracking?) to > SNAT/DNAT packets to/from private network ranges which are not local lan > range > > 2 ? (appears simplest) Using channel variables (addresses) from SIP > messaging to alter addresses prior to bridge. > > > > Any thoughts? > > > > Thanks; > > Bill Ross > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/d6f61011/attachment.html From bk1379 at yahoo.com Tue Nov 11 20:59:07 2014 From: bk1379 at yahoo.com (babak) Date: Tue, 11 Nov 2014 09:59:07 -0800 Subject: [Freeswitch-users] Freeswitch as load balancer in media proxy mode In-Reply-To: <1415508619.56507.YahooMailNeo@web141501.mail.bf1.yahoo.com> References: <1415508619.56507.YahooMailNeo@web141501.mail.bf1.yahoo.com> Message-ID: <1415728747.98527.YahooMailNeo@web141505.mail.bf1.yahoo.com> sorry I meant freeswitch in bypass_media as loadbalancer... is there any comment? ________________________________ Hi I want to use a freeswitch in media proxy mode as load balancer in front of two other freeswitch servers. Is there any sample configuration guide? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/0eef935a/attachment.html From lists at telefaks.de Tue Nov 11 21:37:04 2014 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 11 Nov 2014 19:37:04 +0100 Subject: [Freeswitch-users] Freeswitch as load balancer in media proxy mode In-Reply-To: <1415728747.98527.YahooMailNeo@web141505.mail.bf1.yahoo.com> References: <1415508619.56507.YahooMailNeo@web141501.mail.bf1.yahoo.com> <1415728747.98527.YahooMailNeo@web141505.mail.bf1.yahoo.com> Message-ID: <54625750.6040800@telefaks.de> Hello, sure you can do this. It's best to configure an inbound gateway on one profile (e.g. external) and some outbound gateways to the internal balanced freeswitch servers on onother profile (e.g. internal). The problem is still: How do you balance? By round-robin or by calculating the number of calls on the target server and then forward the call to the server with the least load. Or by time (odd/even seconds)? You may also consider to generate the dialplan via xml-curl (that's how we do it): https://wiki.freeswitch.org/wiki/Mod_xml_curl or get values from memcache (via an external app or the balanced freeswitches' dialplans) and react on it in your load balancer dialplan. https://wiki.freeswitch.org/wiki/Mod_memcache If you have an internal DNS server, you can also set it up to balance the call by round robin. Best regards Peter On 11/11/14 18:59, babak wrote: > > sorry I meant freeswitch in bypass_media as loadbalancer... > is there any comment? > ------------------------------------------------------------------------ > > > Hi > I want to use a freeswitch in media proxy mode as load balancer in > front of two > other freeswitch servers. > Is there any sample configuration guide? > > Regards > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/31990ef3/attachment.html From smrdoshi at gmail.com Tue Nov 11 22:08:21 2014 From: smrdoshi at gmail.com (Samir Doshi) Date: Wed, 12 Nov 2014 00:38:21 +0530 Subject: [Freeswitch-users] Freeswitch as load balancer in media proxy mode In-Reply-To: <54625750.6040800@telefaks.de> References: <1415508619.56507.YahooMailNeo@web141501.mail.bf1.yahoo.com> <1415728747.98527.YahooMailNeo@web141505.mail.bf1.yahoo.com> <54625750.6040800@telefaks.de> Message-ID: Hi, Maybe you can look at Mod_distributor module. https://wiki.freeswitch.org/wiki/Mod_distributor Thanks, Samir On Wed, Nov 12, 2014 at 12:07 AM, Peter Steinbach wrote: > Hello, > > sure you can do this. It's best to configure an inbound gateway on one > profile (e.g. external) and some outbound gateways to the internal balanced > freeswitch servers on onother profile (e.g. internal). > The problem is still: How do you balance? By round-robin or by calculating > the number of calls on the target server and then forward the call to the > server with the least load. Or by time (odd/even seconds)? > > You may also consider to generate the dialplan via xml-curl (that's how we > do it): > https://wiki.freeswitch.org/wiki/Mod_xml_curl > or get values from memcache (via an external app or the balanced > freeswitches' dialplans) and react on it in your load balancer dialplan. > https://wiki.freeswitch.org/wiki/Mod_memcache > > If you have an internal DNS server, you can also set it up to balance the > call by round robin. > > Best regards > Peter > > On 11/11/14 18:59, babak wrote: > > > sorry I meant freeswitch in bypass_media as loadbalancer... > is there any comment? > ------------------------------ > > > Hi > I want to use a freeswitch in media proxy mode as load balancer in front > of two > other freeswitch servers. > Is there any sample configuration guide? > > Regards > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/84a973ec/attachment-0001.html From anthony.minessale at gmail.com Tue Nov 11 23:56:50 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 11 Nov 2014 14:56:50 -0600 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: References: <5460DEF6.6070204@gmail.com> <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> Message-ID: Look in your console log. They are probably being received and not connected to anything. On Tue, Nov 11, 2014 at 2:17 AM, S?ndor Bal?zs < balazs.sandor at virtual-call-center.hu> wrote: > I call a mobile phone from browser with mod_verto, and I would like to > receive dtmfs in the browser (sent from mobile phone) > Sorry if I was not clear enough. > > ?dv?zlettel: > > S?ndor Bal?zs > > Szoftver fejleszt? > > > > Virtual Call Center > > MUNICH | BUDAPEST > | WARSAW > > > Telefon: +36 1 999 7400 > > Web: www.virtual-call-center.hu > > > > > 2014-11-10 22:48 GMT+01:00 Michael Jerris : > >> and by that you mean you want to receive on the server, send from the >> client? That is built in to mod_verto already, and is functional in the >> demo at webrtc.freeswitch.org >> >> On Nov 10, 2014, at 4:42 PM, S?ndor Bal?zs < >> balazs.sandor at virtual-call-center.hu> wrote: >> >> Receive >> On 10 Nov 2014 17:17, "Igor Olhovskiy" wrote: >> >>> You want to receive or send dtmf via mod_verto? >>> >>> On 10.11.14 17:04, Bal?zs S?ndor wrote: >>> >>> Hi all! >>> >>> I would like to receive dtmf with mod_verto. >>> How can I do that? >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/2b08331a/attachment.html From steveayre at gmail.com Wed Nov 12 00:12:04 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 11 Nov 2014 21:12:04 +0000 Subject: [Freeswitch-users] what can I have in ${} construction In-Reply-To: <000001499f0d0fd8-7351ccf4-4547-463f-900e-6d2ebe2c565f-000000@email.amazonses.com> References: <30097.1355410423@ccs.covici.com> <000001499f0d0fd8-7351ccf4-4547-463f-900e-6d2ebe2c565f-000000@email.amazonses.com> Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_commands#mod_commands-FromtheDialplan https://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan There are also examples scattered throughout the wiki/confluence It is tucked away somewhere obscure (after all not all api commands are in mod_commands so it's more generic). Do you have an suggestions for a better location? Seeing Italo's comment it'll be documented in the XML dialplan too which I completely agree is appropriate, but that's still not the only place where it may be used (can also be used via ESL for example) so it may need mentioning elsewhere too. Perhaps a page dedicated to the ${} syntax in all its forms? On 11 November 2014 13:31, Avi Marcus wrote: > Hi - where on confluence is this useful information documented? When > searching, I couldn't find even find it on the wiki. > -Avi > > On Thu, Dec 13, 2012 at 7:45 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> It supports the substring with positive and negative vals so: >> >> ${foo:3:4} would start at char 3 in the string and eval to the next 4 >> chars >> ${foo:-3:2} would start at the end and go back 3 chars then print the >> next 2 chars >> >> If the var name is followed by a ( or a space, it will pass the values to >> the FSAPI and expand the result inline. >> >> ${sofia_contact 1004} >> >> equiv of... >> >> ${sofia_contact(1004)} >> >> >> >> >> >> >> >> On Thu, Dec 13, 2012 at 8:53 AM, wrote: >> >>> Hi. I was experimenting with ${variable} and was wondering what could >>> be in there before the }? I can do a substring if I say variable:2, but >>> I wonder what else you can do? The bash manual has all kinds of things >>> you can have, but how much of this will fs do? >>> >>> 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 >>> covici at ccs.covici.com >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/c4ddbd11/attachment.html From fmyhr at fhmtech.com Wed Nov 12 00:08:21 2014 From: fmyhr at fhmtech.com (Frank Myhr) Date: Tue, 11 Nov 2014 16:08:21 -0500 Subject: [Freeswitch-users] Action in nested condition gets executed even when outer condition is false Message-ID: <54627AC5.2090409@fhmtech.com> Hi, I have the following extension in a context that gets parsed several times during a call due to transfers and execute_extension. I'm finding that one of the inner actions (the correct one, based on time of day) gets executed even when the outer condition is false. Might not have noticed, except my original version of this extension actually played the greeting files, rather than just exporting the correct one in a variable. Callers were getting greeted multiple times, which was not the intention. For now I've sidestepped the audible problem by playing ${greeting} in another extension that does execute_extension on this one. But the greeting is still getting set multiple times per call. Would appreciate any clues as to why an inner action gets executed even though the outer condition is false. Thanks! Frank From steveayre at gmail.com Wed Nov 12 00:16:11 2014 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 11 Nov 2014 21:16:11 +0000 Subject: [Freeswitch-users] setting recording codec In-Reply-To: References: <107291DC4A8A424280FDB589D568639B@gmail.com> Message-ID: There are record_ variables that let you set metadata in recordings (title etc), stereo,rate,direction etc. I would assume it'll be something similar. On 11 November 2014 14:46, Christopher Rienzo wrote: > Getting the lib to do it is easy. It's mainly an API problem- how will FS > expose something like this? > > On Tue, Nov 11, 2014 at 9:38 AM, Brian West wrote: > >> Looking at the libsndfile API this should be doable but I would need to >> read up on it more, If you require this feature email >> consulting at freeswitch.org for a quote on adding this functionality. >> >> Thanks >> >> On Tue, Nov 11, 2014 at 8:33 AM, Brian West wrote: >> >>> If he wants to record without transcoding it is the only viable way to >>> do so currently via mod_native_file. The task was a nice exercise as it >>> exposed a nice little segfault in the stop record. (FS-6980). >>> >>> I fully understood what he wanted, and that it wasn't possible, offering >>> up what we currently can do with mod_native_file. >>> >>> >>> On Tue, Nov 11, 2014 at 7:49 AM, Ben Langfeld >>> wrote: >>> >>>> Brian, was this answer supposed to be to some other question? The >>>> question here was about transcoding the recording, not about stereo vs two >>>> mono recordings... >>>> >>>> On 11 November 2014 11:35, Brian West wrote: >>>> >>>>> just record to a file without any extension. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> "/tmp/${uuid}-testing"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-in.G722] 16000hz >>>>> >>>>> 2014-11-11 07:32:26.713403 [INFO] mod_native_file.c:101 Opening File >>>>> [/tmp/6f1e83a6-1d06-4e31-b641-7d4f825c99d6-testing-out.G722] 16000hz >>>>> >>>>> Now you'll be all set to waste more time dealing with two files vs the >>>>> single wav file. >>>>> >>>>> On Tue, Nov 11, 2014 at 4:33 AM, ik wrote: >>>>> >>>>>> In this specific case, I'm using g711, but can have either a-law or >>>>>> u-law but the service that requires the recording understand only a-law, at >>>>>> the moment I'm using ffmpeg to convert after the call ended, and then send >>>>>> it to that service. >>>>>> >>>>>> Thanks >>>>>> Ido >>>>>> On Nov 11, 2014 12:24 PM, "Stanislav Sinyagin" >>>>>> wrote: >>>>>> >>>>>>> as far as I understand, FreeSWITCH chooses the WAV parameters which >>>>>>> are matching the current channel sampling frequency and codec. For >>>>>>> example, if you record a G722 call, you would get a 16kHZ WAV file. >>>>>>> >>>>>>> Forcing it into one specific encoding would add real-time CPU load. I >>>>>>> think it's still more preferable to run a post-processing job in low >>>>>>> priority -- this way you ensure that the ongoing calls get the best >>>>>>> serving. >>>>>>> >>>>>>> You can easily catch the event of call ending and trigger the >>>>>>> conversion job, by listening to the events via ESL connection. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Nov 11, 2014 at 10:52 AM, ik wrote: >>>>>>> > Thinking in learning better the source code, and might create my >>>>>>> first patch >>>>>>> > to FS :) >>>>>>> > >>>>>>> > On Tue, Nov 11, 2014 at 8:09 AM, Seven Du >>>>>>> wrote: >>>>>>> >> >>>>>>> >> No. Maybe bounty it? >>>>>>> >> >>>>>>> >> On Monday, November 10, 2014 at 8:53 PM, ik wrote: >>>>>>> >> >>>>>>> >> Hello, >>>>>>> >> >>>>>>> >> I could not find any documentation on this subject. >>>>>>> >> I require to have a recording of an a-law wav file (RIFF >>>>>>> (little-endian) >>>>>>> >> data, WAVE audio, ITU G.711 A-law, mono 8000 Hz). >>>>>>> >> >>>>>>> >> Can I set a variable that tells the record command what type of >>>>>>> codec to >>>>>>> >> store the file with ? >>>>>>> >> >>>>>>> >> I know how to convert a normal wav recording into it using ffmpeg >>>>>>> (for >>>>>>> >> example): >>>>>>> >> >>>>>>> >> $ ffmpeg -i 0.wav -ar 8000 -ac 1 -acodec pcm_alaw 0-alaw.wav >>>>>>> >> >>>>>>> >> But I wish to avoid it, if I can use the record app, or use the >>>>>>> >> uuid_record API for it. >>>>>>> >> >>>>>>> >> Thanks, >>>>>>> >> Ido >>>>>>> >> >>>>>>> _________________________________________________________________________ >>>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>>> >> consulting at freeswitch.org >>>>>>> >> http://www.freeswitchsolutions.com >>>>>>> >> >>>>>>> >> Official FreeSWITCH Sites >>>>>>> >> http://www.freeswitch.org >>>>>>> >> http://confluence.freeswitch.org >>>>>>> >> http://www.cluecon.com >>>>>>> >> >>>>>>> >> FreeSWITCH-users mailing list >>>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >> http://www.freeswitch.org >>>>>>> >> >>>>>>> >> >>>>>>> >> >>>>>>> >> >>>>>>> _________________________________________________________________________ >>>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>>> >> consulting at freeswitch.org >>>>>>> >> http://www.freeswitchsolutions.com >>>>>>> >> >>>>>>> >> Official FreeSWITCH Sites >>>>>>> >> http://www.freeswitch.org >>>>>>> >> http://confluence.freeswitch.org >>>>>>> >> http://www.cluecon.com >>>>>>> >> >>>>>>> >> FreeSWITCH-users mailing list >>>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >> http://www.freeswitch.org >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> _________________________________________________________________________ >>>>>>> > Professional FreeSWITCH Consulting Services: >>>>>>> > consulting at freeswitch.org >>>>>>> > http://www.freeswitchsolutions.com >>>>>>> > >>>>>>> > Official FreeSWITCH Sites >>>>>>> > http://www.freeswitch.org >>>>>>> > http://confluence.freeswitch.org >>>>>>> > http://www.cluecon.com >>>>>>> > >>>>>>> > FreeSWITCH-users mailing list >>>>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> > UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> > http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://confluence.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://confluence.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/37ecdf08/attachment-0001.html From msc at freeswitch.org Wed Nov 12 02:13:59 2014 From: msc at freeswitch.org (Michael Collins) Date: Tue, 11 Nov 2014 15:13:59 -0800 Subject: [Freeswitch-users] troubles with regex in mod_ivr? In-Reply-To: <5460DED7.3000702@ringme.ru> References: <5460DED7.3000702@ringme.ru> Message-ID: Test to confirm but if no digits are dialed and the IVR times out then it should move on in the dialplan. You can have the 'exit' action just be the next thing in the dp. -MC On Mon, Nov 10, 2014 at 7:50 AM, ????? wrote: > We need action on timeout, made digits = "/^$/" - nothing. > How to do it correctly? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/2f12482e/attachment.html From msc at freeswitch.org Wed Nov 12 04:29:48 2014 From: msc at freeswitch.org (Michael Collins) Date: Tue, 11 Nov 2014 17:29:48 -0800 Subject: [Freeswitch-users] Action in nested condition gets executed even when outer condition is false In-Reply-To: <54627AC5.2090409@fhmtech.com> References: <54627AC5.2090409@fhmtech.com> Message-ID: I haven't used nested conditions since I was raised in an era when you couldn't nest them. ;) This page mentions the "require-nested" attribute: https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Conditions I didn't see that in your dialplan, so that might be a good place to start. -MC On Tue, Nov 11, 2014 at 1:08 PM, Frank Myhr wrote: > Hi, > > I have the following extension in a context that gets parsed several times > during a call due to transfers and execute_extension. I'm finding > that one of the inner actions (the correct one, based on time of day) gets > executed even when the outer condition is false. Might not have > noticed, except my original version of this extension actually played the > greeting files, rather than just exporting the correct one in a > variable. Callers were getting greeted multiple times, which was not the > intention. > > For now I've sidestepped the audible problem by playing ${greeting} in > another extension that does execute_extension on this one. But the > greeting is still getting set multiple times per call. > > Would appreciate any clues as to why an inner action gets executed even > though the outer condition is false. > > Thanks! > Frank > > > > expression="^set_greeting_tod$"> > > > data="greeting=ivr/ivr-hello.wav"/> > > > > data="greeting=ivr/ivr-good_morning.wav"/> > > > > data="greeting=ivr/ivr-hello.wav"/> > > > > data="greeting=ivr/ivr-good_afternoon.wav"/> > > > > data="greeting=ivr/ivr-good_evening.wav"/> > > > > data="greeting=ivr/ivr-hello.wav"/> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141111/e2d2d060/attachment.html From eduardo.alonso at quobis.com Wed Nov 12 10:48:53 2014 From: eduardo.alonso at quobis.com (Eduardo Alonso) Date: Wed, 12 Nov 2014 08:48:53 +0100 Subject: [Freeswitch-users] Core dump in FreeSWITCH Version 1.2.24+git~20141016T210433Z~30a950b5a9~64bit In-Reply-To: References: Message-ID: Hello Ken: Sorry for my mistake. As I said in my first email, I attached the backtrace file to the JIRA ticket. Thanks for you support. BR. 2014-11-11 17:31 GMT+01:00 Ken Rice : > Please see https://freeswitch.org/confluence/display/FREESWITCH/Debugging > for getting a backtrace. > > Opening a Jira and attach the backtrace. Please do not attach the core > file. > > > On 11/11/14 9:30 AM, "Eduardo Alonso" wrote: > > Hello Brian: > > Thanks for you quick answer. > I downgraded from 1.4 to 1.2 because the error occurred. > > I'm to open a ticket in JIRA with a core-dump file from 1.4. > > Thanks for your support. > BR. > > 2014-11-11 14:25 GMT+01:00 Brian West : > > The mailing list is not the correct place to report bugs, JIRA is, sadly > 1.2.x is EOL and no longer supported. Is this an issue you can replicate > on demand? > > On Tue, Nov 11, 2014 at 5:33 AM, Eduardo Alonso > wrote: > > Dear list: > > This is my first mail to the list. > I hope that my data and email are well formed. > > We are using freeswitch as voicemail in a production environment and every > week we obtain a core dump like the attached file. > > The error occurs in the CoreSession::recordFile function when the customer > try to record a greeting message. > > As is normal it's not possible to attach the entire core file, as well I > attached the backtrace following the debugging guide in the wiki: > https://wiki.freeswitch.org/wiki/Debugging_Freeswitch. > I hope that the attached file help you to find the error. If more > information or testing is needed please let me know. > I'm looking for a similar core in the list, but all errors related with > this issue are different, from my point of view. > > Thank you in advance for your support. > > Cheers and best regards. > > > -- > Ken > > > > *http://www.FreeSWITCH.org > http://www.ClueCon.com http://www.OSTAG.org > *irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Eduardo Alonso Gil* VoIP Systems Engineer @ Quobis | e: eduardo.alonso at quobis.com | t: +34902999465 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/e6e31ec9/attachment.html From bordmi at rarus.ru Wed Nov 12 08:36:03 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Wed, 12 Nov 2014 09:36:03 +0400 Subject: [Freeswitch-users] troubles with regex in mod_ivr? In-Reply-To: References: <5460DED7.3000702@ringme.ru> Message-ID: >From the FreeSWITCH Wiki ( https://wiki.freeswitch.org/wiki/IVR_Menu#How_to_route_the_call_if_no_DTMF_is_pressed ): How to route the call if no DTMF is pressed First, define the IVR main menu like this: and then in your dialplan, you need: With this config, if the IVR manages to bridge the call, it will hang up when the bridge ends. But if no DTMF is sent, twice in a row (max-timeouts control that), FreeSWITCH will exit the IVR menu and process the next dialplan line, which bridges to 1000. 2014-11-12 2:13 GMT+03:00 Michael Collins : > Test to confirm but if no digits are dialed and the IVR times out then it > should move on in the dialplan. You can have the 'exit' action just be the > next thing in the dp. > -MC > > On Mon, Nov 10, 2014 at 7:50 AM, ????? wrote: > >> We need action on timeout, made digits = "/^$/" - nothing. >> How to do it correctly? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ? ?????????, ??????? ??????? ????????????? ?? ?????????? VoIP ??????? ????????? ????????????? ?????? ???????? ?????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/c78bb8c6/attachment-0001.html From karl-theo_hofer at inteli-sim.com Wed Nov 12 15:20:01 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Wed, 12 Nov 2014 13:20:01 +0100 Subject: [Freeswitch-users] no ring back tone In-Reply-To: References: <54613814.3060504@inteli-sim.com> Message-ID: <54635071.8070205@inteli-sim.com> Hi Guys thanks for your answers but I had done this already Thats the originate for the first leg $bridge = $con->bgapi("originate", "{GT_calling_address=$FS_A_calling_address,GT_called_address=$FS_A_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,presence_data='$GT_callid $a_legid',GT_callid=$GT_callid,call_leg=A,return_ring_ready=true,ignore_early_media=false,bridge_early_media=true,GT_Start_time=$a_GT_start_time,GT_legid=$a_legid,GT_deskcall=yes,originate_timeout=$FS_A_timeout,origination_caller_id_number=$FS_A_calling_address,origination_caller_id_name=$FS_A_calling_address,effective_caller_id_number=$FS_A_calling_address,effective_caller_id_name=$FS_A_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:GATEWAY_DOWN:INVALID_GATEWAY:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE:USER_BUSY}$a_local_complete_dialstring &park()"); And here for the second leg $con->bgapi("originate", "{GT_calling_address=$FS_B_calling_address,GT_called_address=$FS_B_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,originate_timeout=$FS_B_timeout,presence_data='$GT_callid $b_legid',instant_ringback=false,GT_callid=$GT_callid,A_uuid=$leg_uuid,call_leg=B,return_ring_ready=yes,bridge_early_media=true,ignore_early_media=false,GT_Start_time=$b_GT_start_time,GT_legid=$b_legid,GT_deskcall=yes,origination_caller_id_number=$FS_B_calling_address,origination_caller_id_name=$FS_B_calling_address,effective_caller_id_name=$FS_B_calling_address,effective_caller_id_number=$FS_B_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:INVALID_GATEWAY:GATEWAY_DOWN:NORMAL_TEMPORARY_FAILURE:NO_ANSWER:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE}$b_local_complete_dialstring &park()"); An how i see this I have set the right parameters but still no RBT and on both side I get 183 SDP back from the carrier and it includes RBT I really don't get it why i do not have the RBT from B on the A side. With best regards Karl Theo Hofer lakshmanan ganapathy skrev 2014-11-11 05:20: > I think you need to set ignore_early_media=false and > bridge_early_media=true, which will pass the ringback tone from B leg to A > leg. > > > > On Tue, Nov 11, 2014 at 3:41 AM, kthofer > wrote: > >> Hi there >> >> please be gentle with me but i can not find any help in the wiki to >> solve my problem >> this is a desperat try to get some answers to the following problem >> freeswitch is a configured as call back server >> FS originates the A-leg when answered the B-leg is originated. >> after A-leg has answered I like to play a short welcome message >> After the welcome mesage is played the international, original >> ringbacktone (telco carrier RBT) from the B-leg shall be passed through >> and the A-leg user should be able to hear it. >> This is important to hear some carrier specific announcements or >> coloured/personalized ring back. >> Right now we play our own ringbacktone even though the carrier provides >> us with a 183 sdp >> I did play with the usual Channel variables >> but as far as I understood the passing through of the ringbacktone is >> default behaviour. >> >> >> >> >> >> >> >> >> -- >> With best regards >> kT >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> I think you need to set ignore_early_media=false and >> bridge_early_media=true, which will pass the ringback tone from B leg >> to A leg. >> >> >> >> On Tue, Nov 11, 2014 at 3:41 AM, kthofer >> > > wrote: >> >> Hi there >> >> please be gentle with me but i can not find any help in the wiki to >> solve my problem >> this is a desperat try to get some answers to the following problem >> freeswitch is a configured as call back server >> FS originates the A-leg when answered the B-leg is originated. >> after A-leg has answered I like to play a short welcome message >> After the welcome mesage is played the international, original >> ringbacktone (telco carrier RBT) from the B-leg shall be passed >> through >> and the A-leg user should be able to hear it. >> This is important to hear some carrier specific announcements or >> coloured/personalized ring back. >> Right now we play our own ringbacktone even though the carrier >> provides >> us with a 183 sdp >> I did play with the usual Channel variables >> but as far as I understood the passing through of the ringbacktone is >> default behaviour. >> >> >> >> >> >> >> >> >> -- >> With best regards >> kT >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/402c39f2/attachment.html From nick.zaitsev at mail.ru Wed Nov 12 15:24:02 2014 From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=) Date: Wed, 12 Nov 2014 15:24:02 +0300 Subject: [Freeswitch-users] =?utf-8?q?Skinny_firmware_version?= Message-ID: <1415795042.761414668@f300.i.mail.ru> Good day to you. I try to set the Cisco 7985 to work with freeswitch. But i have a problem: 7985 requests the firmware version 2014-11-12 14:23:41.660763 [WARNING] skinny_server.c:1762 Device SEP0050600361BC:1 is requesting for firmware version, but none is set. 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:1767 [SEP0050600361BC:1 @ 10.4.33.12:27811] Send Version with Version() 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:27811] Received HeadsetStatusMessage (type=2b,length=8). 2014-11-12 14:23:41.660763 [DEBUG] skinny_server.c:2159 [SEP0050600361BC:1 @ 10.4.33.12:27811] Update headset accessory status (OnHook) I've put into the config file. but it is still request for the firmware.(I supposed that there is not any difference,which firmware version to set).Moreover, there is "Configuring IP" string on the phone display,however the phone works and i can make calls. Please,give me an advice. Thank you for your time, Best regards. -- Nick Zaitsev ---------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/6ae74e33/attachment.html From steveayre at gmail.com Wed Nov 12 16:50:27 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 12 Nov 2014 13:50:27 +0000 Subject: [Freeswitch-users] no ring back tone In-Reply-To: <54635071.8070205@inteli-sim.com> References: <54613814.3060504@inteli-sim.com> <54635071.8070205@inteli-sim.com> Message-ID: >From what you've posted there they're two separate unconnected calls. So there's no reason one would hear the other. You need to bridge them together, and in time for them to hear the early media. On 12 November 2014 12:20, kthofer wrote: > Hi Guys > > thanks for your answers > but I had done this already > > Thats the originate for the first leg > > $bridge = $con->bgapi("originate", > "{GT_calling_address=$FS_A_calling_address,GT_called_address=$FS_A_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,presence_data='$GT_callid > $a_legid',GT_callid=$GT_callid,call_leg=A,return_ring_ready=true,ignore_early_media=false,bridge_early_media=true,GT_Start_time=$a_GT_start_time,GT_legid=$a_legid,GT_deskcall=yes,originate_timeout=$FS_A_timeout,origination_caller_id_number=$FS_A_calling_address,origination_caller_id_name=$FS_A_calling_address,effective_caller_id_number=$FS_A_calling_address,effective_caller_id_name=$FS_A_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:GATEWAY_DOWN:INVALID_GATEWAY:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE:USER_BUSY}$a_local_complete_dialstring > &park()"); > > > And here for the second leg > $con->bgapi("originate", > "{GT_calling_address=$FS_B_calling_address,GT_called_address=$FS_B_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,originate_timeout=$FS_B_timeout,presence_data='$GT_callid > $b_legid',instant_ringback=false,GT_callid=$GT_callid,A_uuid=$leg_uuid,call_leg=B,return_ring_ready=yes,bridge_early_media=true,ignore_early_media=false,GT_Start_time=$b_GT_start_time,GT_legid=$b_legid,GT_deskcall=yes,origination_caller_id_number=$FS_B_calling_address,origination_caller_id_name=$FS_B_calling_address,effective_caller_id_name=$FS_B_calling_address,effective_caller_id_number=$FS_B_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:INVALID_GATEWAY:GATEWAY_DOWN:NORMAL_TEMPORARY_FAILURE:NO_ANSWER:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE}$b_local_complete_dialstring > &park()"); > > > An how i see this I have set the right parameters but still no RBT > and on both side I get 183 SDP back from the carrier and it includes RBT > > I really don't get it why i do not have the RBT from B on the A side. > > With best regards > > Karl Theo Hofer > > > lakshmanan ganapathy skrev 2014-11-11 05:20: > > I think you need to set ignore_early_media=false and > bridge_early_media=true, which will pass the ringback tone from B leg to A > leg. > > > > On Tue, Nov 11, 2014 at 3:41 AM, kthofer > wrote: > > > Hi there > > please be gentle with me but i can not find any help in the wiki to > solve my problem > this is a desperat try to get some answers to the following problem > freeswitch is a configured as call back server > FS originates the A-leg when answered the B-leg is originated. > after A-leg has answered I like to play a short welcome message > After the welcome mesage is played the international, original > ringbacktone (telco carrier RBT) from the B-leg shall be passed through > and the A-leg user should be able to hear it. > This is important to hear some carrier specific announcements or > coloured/personalized ring back. > Right now we play our own ringbacktone even though the carrier provides > us with a 183 sdp > I did play with the usual Channel variables > but as far as I understood the passing through of the ringbacktone is > default behaviour. > > > > > > > > > -- > With best regards > kT > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > I think you need to set ignore_early_media=false and > bridge_early_media=true, which will pass the ringback tone from B leg to A > leg. > > > > On Tue, Nov 11, 2014 at 3:41 AM, kthofer > wrote: > >> Hi there >> >> please be gentle with me but i can not find any help in the wiki to >> solve my problem >> this is a desperat try to get some answers to the following problem >> freeswitch is a configured as call back server >> FS originates the A-leg when answered the B-leg is originated. >> after A-leg has answered I like to play a short welcome message >> After the welcome mesage is played the international, original >> ringbacktone (telco carrier RBT) from the B-leg shall be passed through >> and the A-leg user should be able to hear it. >> This is important to hear some carrier specific announcements or >> coloured/personalized ring back. >> Right now we play our own ringbacktone even though the carrier provides >> us with a 183 sdp >> I did play with the usual Channel variables >> but as far as I understood the passing through of the ringbacktone is >> default behaviour. >> >> >> >> >> >> >> >> >> -- >> With best regards >> kT >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/4447fdf9/attachment-0001.html From steveayre at gmail.com Wed Nov 12 16:54:32 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 12 Nov 2014 13:54:32 +0000 Subject: [Freeswitch-users] Action in nested condition gets executed even when outer condition is false In-Reply-To: References: <54627AC5.2090409@fhmtech.com> Message-ID: require_nested defaults to true so shouldn't be required. Frank, collect a debug-level log of the call. It'll show the parsing of the dialplan, and might shed light on what's happening. On 12 November 2014 01:29, Michael Collins wrote: > I haven't used nested conditions since I was raised in an era when you > couldn't nest them. ;) > > This page mentions the "require-nested" attribute: > > https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Conditions > > I didn't see that in your dialplan, so that might be a good place to start. > > -MC > > > On Tue, Nov 11, 2014 at 1:08 PM, Frank Myhr wrote: > >> Hi, >> >> I have the following extension in a context that gets parsed several >> times during a call due to transfers and execute_extension. I'm finding >> that one of the inner actions (the correct one, based on time of day) >> gets executed even when the outer condition is false. Might not have >> noticed, except my original version of this extension actually played the >> greeting files, rather than just exporting the correct one in a >> variable. Callers were getting greeted multiple times, which was not the >> intention. >> >> For now I've sidestepped the audible problem by playing ${greeting} in >> another extension that does execute_extension on this one. But the >> greeting is still getting set multiple times per call. >> >> Would appreciate any clues as to why an inner action gets executed even >> though the outer condition is false. >> >> Thanks! >> Frank >> >> >> >> > expression="^set_greeting_tod$"> >> >> >> > data="greeting=ivr/ivr-hello.wav"/> >> >> >> >> > data="greeting=ivr/ivr-good_morning.wav"/> >> >> >> >> > data="greeting=ivr/ivr-hello.wav"/> >> >> >> >> > data="greeting=ivr/ivr-good_afternoon.wav"/> >> >> >> >> > data="greeting=ivr/ivr-good_evening.wav"/> >> >> >> >> > data="greeting=ivr/ivr-hello.wav"/> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/e90c2c7a/attachment.html From steveayre at gmail.com Wed Nov 12 16:57:01 2014 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 12 Nov 2014 13:57:01 +0000 Subject: [Freeswitch-users] Action in nested condition gets executed even when outer condition is false In-Reply-To: References: <54627AC5.2090409@fhmtech.com> Message-ID: Worth noting you don't need nested conditions to do this - search for Time-of-day-tod on https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan On 12 November 2014 13:54, Steven Ayre wrote: > require_nested defaults to true so shouldn't be required. > > Frank, collect a debug-level log of the call. It'll show the parsing of > the dialplan, and might shed light on what's happening. > > On 12 November 2014 01:29, Michael Collins wrote: > >> I haven't used nested conditions since I was raised in an era when you >> couldn't nest them. ;) >> >> This page mentions the "require-nested" attribute: >> >> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Conditions >> >> I didn't see that in your dialplan, so that might be a good place to >> start. >> >> -MC >> >> >> On Tue, Nov 11, 2014 at 1:08 PM, Frank Myhr wrote: >> >>> Hi, >>> >>> I have the following extension in a context that gets parsed several >>> times during a call due to transfers and execute_extension. I'm finding >>> that one of the inner actions (the correct one, based on time of day) >>> gets executed even when the outer condition is false. Might not have >>> noticed, except my original version of this extension actually played >>> the greeting files, rather than just exporting the correct one in a >>> variable. Callers were getting greeted multiple times, which was not the >>> intention. >>> >>> For now I've sidestepped the audible problem by playing ${greeting} in >>> another extension that does execute_extension on this one. But the >>> greeting is still getting set multiple times per call. >>> >>> Would appreciate any clues as to why an inner action gets executed even >>> though the outer condition is false. >>> >>> Thanks! >>> Frank >>> >>> >>> >>> >> expression="^set_greeting_tod$"> >>> >>> >>> >> data="greeting=ivr/ivr-hello.wav"/> >>> >>> >>> >>> >> data="greeting=ivr/ivr-good_morning.wav"/> >>> >>> >>> >>> >> data="greeting=ivr/ivr-hello.wav"/> >>> >>> >>> >>> >> data="greeting=ivr/ivr-good_afternoon.wav"/> >>> >>> >>> >> break="on-true"> >>> >> data="greeting=ivr/ivr-good_evening.wav"/> >>> >>> >>> >>> >> data="greeting=ivr/ivr-hello.wav"/> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/e2642986/attachment.html From rossbcan at gmail.com Wed Nov 12 17:35:55 2014 From: rossbcan at gmail.com (Bill Ross) Date: Wed, 12 Nov 2014 09:35:55 -0500 Subject: [Freeswitch-users] FS rewrites RTP outgoing port. Client requires same port as incoming, no RX RTP Message-ID: <008201cffe85$f83f30b0$e8bd9210$@gmail.com> Folks; FS: 1.4.13 / OpenWrt disable-rtp-auto-adjust (default: false) is at default. FS is re-writing RTP to client port. Client requires same RX port as it is sending on. Can FS "somehow" be configured to use same port as incoming RTP? It is my understanding that rtp-auto-adjust only adjusts IP's if foreign and, ports if outside of rtp-start-port / rtp-end-port range Is this correct? Should I report a bug? Regards; Bill Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/bae82a3a/attachment-0001.html From rtreleaven at bunnykick.ca Wed Nov 12 17:56:57 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Wed, 12 Nov 2014 14:56:57 +0000 Subject: [Freeswitch-users] FS rewrites RTP outgoing port. Client requires same port as incoming, no RX RTP In-Reply-To: <008201cffe85$f83f30b0$e8bd9210$@gmail.com> References: <008201cffe85$f83f30b0$e8bd9210$@gmail.com> Message-ID: Someone will surely correct me if I am wrong about this. The default behavior is for freeswitch to accept rtp from any ip address or port during a small time window at the beginning of the call. rtp-start-port / rtp-end-port define the ports freeswitch listens on. rtp-auto-adjust is about freeswitch being permissive about where the rtp is coming from. Hope that is clear Regards, Russell On Wed, Nov 12, 2014 at 2:35 PM, Bill Ross wrote: > Folks; > > FS: 1.4.13 / OpenWrt > > disable-rtp-auto-adjust (default: false) is at default. > > > > FS is re-writing RTP to client port. Client requires same RX port as it is > sending on. > > > > Can FS ?somehow? be configured to use same port as incoming RTP? > > > > It is my understanding that rtp-auto-adjust only adjusts IP?s if foreign > and, ports if outside of rtp-start-port / rtp-end-port range > > Is this correct? > > > > Should I report a bug? > > > > Regards; > > Bill Ross > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/a3ecc18c/attachment.html From brian at freeswitch.org Wed Nov 12 18:13:25 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Nov 2014 09:13:25 -0600 Subject: [Freeswitch-users] FS rewrites RTP outgoing port. Client requires same port as incoming, no RX RTP In-Reply-To: <008201cffe85$f83f30b0$e8bd9210$@gmail.com> References: <008201cffe85$f83f30b0$e8bd9210$@gmail.com> Message-ID: You want auto adjust, but what it sounds like is your don't have static port mappings for RTP ports on your nat, This means that the SDP may say port 12345 ip 5.6.7.8, but the NAT randomizes the port to something else say port 44567 ip 5.6.7.8, In which case during the small window we'll see HEY, looks like this client doesn't know how to traverse nat all the way, we'll switch to that port making it work. If you're unable to get NAT working then you have to wonder if you have an ALG also, Happen to have a SIP trace of this? On Wed, Nov 12, 2014 at 8:35 AM, Bill Ross wrote: > Folks; > > FS: 1.4.13 / OpenWrt > > disable-rtp-auto-adjust (default: false) is at default. > > > > FS is re-writing RTP to client port. Client requires same RX port as it is > sending on. > > > > Can FS ?somehow? be configured to use same port as incoming RTP? > > > > It is my understanding that rtp-auto-adjust only adjusts IP?s if foreign > and, ports if outside of rtp-start-port / rtp-end-port range > > Is this correct? > > > > Should I report a bug? > > > > Regards; > > Bill Ross > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/4dd762b5/attachment.html From karl-theo_hofer at inteli-sim.com Wed Nov 12 18:56:05 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Wed, 12 Nov 2014 16:56:05 +0100 Subject: [Freeswitch-users] no ring back tone In-Reply-To: References: <54613814.3060504@inteli-sim.com> <54635071.8070205@inteli-sim.com> Message-ID: <54638315.70102@inteli-sim.com> Hi Steven of course i bridge the channels see below. I am really puzzled. here the second leg inclusive the bridge command $con->bgapi("originate", "{GT_calling_address=$FS_B_calling_address,GT_called_address=$FS_B_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,originate_timeout=$FS_B_timeout,presence_data='$GT_callid $b_legid',instant_ringback=false,GT_callid=$GT_callid,A_uuid=$leg_uuid,call_leg=B,return_ring_ready=yes,bridge_early_media=true,ignore_early_media=false,GT_Start_time=$b_GT_start_time,GT_legid=$b_legid,GT_deskcall=yes,origination_caller_id_number=$FS_B_calling_address,origination_caller_id_name=$FS_B_calling_address,effective_caller_id_name=$FS_B_calling_address,effective_caller_id_number=$FS_B_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:INVALID_GATEWAY:GATEWAY_DOWN:NORMAL_TEMPORARY_FAILURE:NO_ANSWER:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE}$b_local_complete_dialstring &park()"); } else { freeswitch::console_log("info", "NOT SEQUENCIAL\n"); } } elsif($ab_leg eq 'B') { freeswitch::console_log("info", "B answered $GT_callid\n"); $B_connected_uuid = $leg_uuid; $B_GT_connected_GW = $sipgw; $B_GT_connect_time = $localtime; $B_GT_GATEWAYS{$sipgw}{"$sipgw\_connect_time"} = "$localtime"; if($FS_call_type eq 'sequential') { my $A_uuid = extract_e_header($event, "variable_A_uuid"); $con->bgapi("uuid_bridge", "$A_uuid $leg_uuid"); $have_bridge = 'yes'; With best regards Karl Theo Hofer Steven Ayre skrev 2014-11-12 14:50: > >From what you've posted there they're two separate unconnected calls. So > there's no reason one would hear the other. You need to bridge them > together, and in time for them to hear the early media. > > On 12 November 2014 12:20, kthofer wrote: > >> Hi Guys >> >> thanks for your answers >> but I had done this already >> >> Thats the originate for the first leg >> >> $bridge = $con->bgapi("originate", >> "{GT_calling_address=$FS_A_calling_address,GT_called_address=$FS_A_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,presence_data='$GT_callid >> $a_legid',GT_callid=$GT_callid,call_leg=A,return_ring_ready=true,ignore_early_media=false,bridge_early_media=true,GT_Start_time=$a_GT_start_time,GT_legid=$a_legid,GT_deskcall=yes,originate_timeout=$FS_A_timeout,origination_caller_id_number=$FS_A_calling_address,origination_caller_id_name=$FS_A_calling_address,effective_caller_id_number=$FS_A_calling_address,effective_caller_id_name=$FS_A_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:GATEWAY_DOWN:INVALID_GATEWAY:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE:USER_BUSY}$a_local_complete_dialstring >> &park()"); >> >> >> And here for the second leg >> $con->bgapi("originate", >> "{GT_calling_address=$FS_B_calling_address,GT_called_address=$FS_B_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,originate_timeout=$FS_B_timeout,presence_data='$GT_callid >> $b_legid',instant_ringback=false,GT_callid=$GT_callid,A_uuid=$leg_uuid,call_leg=B,return_ring_ready=yes,bridge_early_media=true,ignore_early_media=false,GT_Start_time=$b_GT_start_time,GT_legid=$b_legid,GT_deskcall=yes,origination_caller_id_number=$FS_B_calling_address,origination_caller_id_name=$FS_B_calling_address,effective_caller_id_name=$FS_B_calling_address,effective_caller_id_number=$FS_B_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:INVALID_GATEWAY:GATEWAY_DOWN:NORMAL_TEMPORARY_FAILURE:NO_ANSWER:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE}$b_local_complete_dialstring >> &park()"); >> >> >> An how i see this I have set the right parameters but still no RBT >> and on both side I get 183 SDP back from the carrier and it includes RBT >> >> I really don't get it why i do not have the RBT from B on the A side. >> >> With best regards >> >> Karl Theo Hofer >> >> >> lakshmanan ganapathy skrev 2014-11-11 05:20: >> >> I think you need to set ignore_early_media=false and >> bridge_early_media=true, which will pass the ringback tone from B leg to A >> leg. >> >> >> >> On Tue, Nov 11, 2014 at 3:41 AM, kthofer >> wrote: >> >> >> Hi there >> >> please be gentle with me but i can not find any help in the wiki to >> solve my problem >> this is a desperat try to get some answers to the following problem >> freeswitch is a configured as call back server >> FS originates the A-leg when answered the B-leg is originated. >> after A-leg has answered I like to play a short welcome message >> After the welcome mesage is played the international, original >> ringbacktone (telco carrier RBT) from the B-leg shall be passed through >> and the A-leg user should be able to hear it. >> This is important to hear some carrier specific announcements or >> coloured/personalized ring back. >> Right now we play our own ringbacktone even though the carrier provides >> us with a 183 sdp >> I did play with the usual Channel variables >> but as far as I understood the passing through of the ringbacktone is >> default behaviour. >> >> >> >> >> >> >> >> >> -- >> With best regards >> kT >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> I think you need to set ignore_early_media=false and >> bridge_early_media=true, which will pass the ringback tone from B leg to A >> leg. >> >> >> >> On Tue, Nov 11, 2014 at 3:41 AM, kthofer >> wrote: >> >>> Hi there >>> >>> please be gentle with me but i can not find any help in the wiki to >>> solve my problem >>> this is a desperat try to get some answers to the following problem >>> freeswitch is a configured as call back server >>> FS originates the A-leg when answered the B-leg is originated. >>> after A-leg has answered I like to play a short welcome message >>> After the welcome mesage is played the international, original >>> ringbacktone (telco carrier RBT) from the B-leg shall be passed through >>> and the A-leg user should be able to hear it. >>> This is important to hear some carrier specific announcements or >>> coloured/personalized ring back. >>> Right now we play our own ringbacktone even though the carrier provides >>> us with a 183 sdp >>> I did play with the usual Channel variables >>> but as far as I understood the passing through of the ringbacktone is >>> default behaviour. >>> >>> >>> >>> >>> >>> >>> >>> >>> -- >>> With best regards >>> kT >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> From what you've posted there they're two separate unconnected calls. >> So there's no reason one would hear the other. You need to bridge >> them together, and in time for them to hear the early media. >> >> On 12 November 2014 12:20, kthofer > > wrote: >> >> Hi Guys >> >> thanks for your answers >> but I had done this already >> >> Thats the originate for the first leg >> >> $bridge = $con->bgapi("originate", >> "{GT_calling_address=$FS_A_calling_address,GT_called_address=$FS_A_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,presence_data='$GT_callid >> $a_legid',GT_callid=$GT_callid,call_leg=A,return_ring_ready=true,ignore_early_media=false,bridge_early_media=true,GT_Start_time=$a_GT_start_time,GT_legid=$a_legid,GT_deskcall=yes,originate_timeout=$FS_A_timeout,origination_caller_id_number=$FS_A_calling_address,origination_caller_id_name=$FS_A_calling_address,effective_caller_id_number=$FS_A_calling_address,effective_caller_id_name=$FS_A_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:GATEWAY_DOWN:INVALID_GATEWAY:NORMAL_TEMPORARY_FAILURE:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE:USER_BUSY}$a_local_complete_dialstring >> &park()"); >> >> >> And here for the second leg >> $con->bgapi("originate", >> "{GT_calling_address=$FS_B_calling_address,GT_called_address=$FS_B_called_address,sip_codec_negotiation=scrooge,uuid_bridge_continue_on_cancel=true,originate_timeout=$FS_B_timeout,presence_data='$GT_callid >> $b_legid',instant_ringback=false,GT_callid=$GT_callid,A_uuid=$leg_uuid,call_leg=B,return_ring_ready=yes,bridge_early_media=true,ignore_early_media=false,GT_Start_time=$b_GT_start_time,GT_legid=$b_legid,GT_deskcall=yes,origination_caller_id_number=$FS_B_calling_address,origination_caller_id_name=$FS_B_calling_address,effective_caller_id_name=$FS_B_calling_address,effective_caller_id_number=$FS_B_calling_address,fail_on_single_reject=ORIGINATOR_CANCEL,continue_on_fail=^^:INVALID_GATEWAY:GATEWAY_DOWN:NORMAL_TEMPORARY_FAILURE:NO_ANSWER:NO_ROUTE_DESTINATION:UNALLOCATED_NUMBER:NO_USER_RESPONSE}$b_local_complete_dialstring >> &park()"); >> >> >> An how i see this I have set the right parameters but still no RBT >> and on both side I get 183 SDP back from the carrier and it >> includes RBT >> >> I really don't get it why i do not have the RBT from B on the A side. >> >> With best regards >> >> Karl Theo Hofer >> >> lakshmanan ganapathy skrev 2014-11-11 05:20: >>> I think you need to set ignore_early_media=false and >>> bridge_early_media=true, which will pass the ringback tone from B leg to A >>> leg. >>> >>> >>> >>> On Tue, Nov 11, 2014 at 3:41 AM, kthofer >>> wrote: >>> >>>> Hi there >>>> >>>> please be gentle with me but i can not find any help in the wiki to >>>> solve my problem >>>> this is a desperat try to get some answers to the following problem >>>> freeswitch is a configured as call back server >>>> FS originates the A-leg when answered the B-leg is originated. >>>> after A-leg has answered I like to play a short welcome message >>>> After the welcome mesage is played the international, original >>>> ringbacktone (telco carrier RBT) from the B-leg shall be passed through >>>> and the A-leg user should be able to hear it. >>>> This is important to hear some carrier specific announcements or >>>> coloured/personalized ring back. >>>> Right now we play our own ringbacktone even though the carrier provides >>>> us with a 183 sdp >>>> I did play with the usual Channel variables >>>> but as far as I understood the passing through of the ringbacktone is >>>> default behaviour. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> With best regards >>>> kT >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> I think you need to set ignore_early_media=false and >>>> bridge_early_media=true, which will pass the ringback tone from >>>> B leg to A leg. >>>> >>>> >>>> >>>> On Tue, Nov 11, 2014 at 3:41 AM, kthofer >>>> >>> > wrote: >>>> >>>> Hi there >>>> >>>> please be gentle with me but i can not find any help in the >>>> wiki to >>>> solve my problem >>>> this is a desperat try to get some answers to the following >>>> problem >>>> freeswitch is a configured as call back server >>>> FS originates the A-leg when answered the B-leg is originated. >>>> after A-leg has answered I like to play a short welcome >>>> message >>>> After the welcome mesage is played the international, original >>>> ringbacktone (telco carrier RBT) from the B-leg shall be >>>> passed through >>>> and the A-leg user should be able to hear it. >>>> This is important to hear some carrier specific >>>> announcements or >>>> coloured/personalized ring back. >>>> Right now we play our own ringbacktone even though the >>>> carrier provides >>>> us with a 183 sdp >>>> I did play with the usual Channel variables >>>> but as far as I understood the passing through of the >>>> ringbacktone is >>>> default behaviour. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> With best regards >>>> kT >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/aa6b54fd/attachment-0001.html From wampir990 at gmail.com Wed Nov 12 19:21:10 2014 From: wampir990 at gmail.com (Jacek) Date: Wed, 12 Nov 2014 17:21:10 +0100 Subject: [Freeswitch-users] How to use mod_skypopen with the new Skype-4.3 without Pulseaudio server? Message-ID: <546388F6.2020703@gmail.com> Hi I read a tutorial by Giovanni Maruzzelli. http://lists.freeswitch.org/pipermail/freeswitch-users/2014-August/107455.html These are my notes: Theoretically, Skype 4.3.x is demanding Pulseaudio audio output. I write theoretically, because I noticed that Skype works without a problem thanks to the ALSA library itself Apulse. https://github.com/i-rinat/apulse In connection with the question whether it is possible with the help of this library to connect Skype with FreeSWITCH using eg mod_alsa or directly mod_skypopen? Skypopen once used the ALSA API (version 1.0.x, as I recall). Under ideal conditions, it would be best to libraries based on Apulse do fork - eg library FsPulse, that sound from Skype headed straight to the driver interface Skyopen, eg / dev / skypopen. Unfortunately, I am not a programmer, and I do not know how to change the ALSA library API to OSS compatible with FreeSWITCH. But if someone could create such a library to integrate Skype - FreeSWITCH, it for many years to solve the problem with Skype, and potential problems with possible errors Pulseaudio, Pulse which are patched often even after two years, such as the famous error: 866 http://lists.freedesktop.org/archives/pulseaudio-bugs/2010-October/004195.html If the cooperation of Skype with FreeSWITCH to be stable and predictable, Pulseaudio in the middle of it is not in my opinion a good idea too. Also would be easier to control operation of Skype by FreeSWITCH, if the sound from Skype went directly to the server, without mediators, where the shape and function of FreeSWITCH developers have no effect. PS My natural language is Polish, sorry for bad English. Yours -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/24425053/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 213 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/24425053/attachment.bin From olegstolyar at gmail.com Wed Nov 12 20:27:38 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Nov 2014 09:27:38 -0800 Subject: [Freeswitch-users] Music on hold sample rate mismatch Message-ID: This might be the easiest question ever asked on this list. I am playing wav files with sample rate 48000 as music on hold for a conference with sample rate 8000. I am getting this warning in the logs: switch_core_file.c:211 File /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't match requested rate 8000 I can ignore it, right? This will not create any actual problems? I don't think it will even increase the CPU consumption but wanted to confirm. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/9904c252/attachment.html From my.post at hotmail.com Wed Nov 12 20:26:20 2014 From: my.post at hotmail.com (Pavel) Date: Wed, 12 Nov 2014 23:26:20 +0600 Subject: [Freeswitch-users] Trying to originate a call using ES. Message-ID: Hello Everybody! I'm stuck trying to originate a call using event socket. Somewhere on the internet I've found some Perl script involving ESL. So I changed it a bit so I get an incoming call to outbound socket, look for destination number and trying to set up outgoing call using originate like so: $con->sendRecv("api originate user/1000 &park()") This works. The question is how can I control a unanswered timeout of this originate, say to playback some message to initial caller after 5 seconds ? I've tried setting the following variables: originate_timeout=5 leg_timeout=5 call_timeout=5 like so: $con->sendRecv("api originate {originate_timeout=5}user/1000 &park()") None of this working. The originated call keeps ringing. Please advise. Thanks. Regards, Pavel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/37a5637b/attachment.html From fmyhr at fhmtech.com Wed Nov 12 20:58:19 2014 From: fmyhr at fhmtech.com (Frank Myhr) Date: Wed, 12 Nov 2014 12:58:19 -0500 Subject: [Freeswitch-users] Action in nested condition gets executed even when outer condition is false In-Reply-To: References: <54627AC5.2090409@fhmtech.com> Message-ID: <54639FBB.8010803@fhmtech.com> Steven and Michael, Thank you very much for your help! I looked at a debug-level call log yesterday before posting here and it only confused me because the outer condition regex FAILed but the inner action still got executed. Steven, thank you VERY much for pointing out that my extension doesn't even need the nested condition. I'd seen the wiki example, but clearly need to slow down and read more carefully! I've re-written the extension using the default break="on-false" for the first condition and no nesting, and it now works as expected. General lessons I take from this: 1) This list rocks! 2) Avoid nested conditions if at all possible. Thanks again! Frank On 11/12/2014 08:57 AM, Steven Ayre wrote: > Worth noting you don't need nested conditions to do this - search > for Time-of-day-tod on > https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan > > On 12 November 2014 13:54, Steven Ayre wrote: > >> require_nested defaults to true so shouldn't be required. >> >> Frank, collect a debug-level log of the call. It'll show the parsing of >> the dialplan, and might shed light on what's happening. >> >> On 12 November 2014 01:29, Michael Collins wrote: >> >>> I haven't used nested conditions since I was raised in an era when you >>> couldn't nest them. ;) >>> >>> This page mentions the "require-nested" attribute: >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/XML+Dialplan#XMLDialplan-Conditions >>> >>> I didn't see that in your dialplan, so that might be a good place to >>> start. >>> >>> -MC >>> >>> >>> On Tue, Nov 11, 2014 at 1:08 PM, Frank Myhr wrote: >>> >>>> Hi, >>>> >>>> I have the following extension in a context that gets parsed several >>>> times during a call due to transfers and execute_extension. I'm finding >>>> that one of the inner actions (the correct one, based on time of day) >>>> gets executed even when the outer condition is false. Might not have >>>> noticed, except my original version of this extension actually played >>>> the greeting files, rather than just exporting the correct one in a >>>> variable. Callers were getting greeted multiple times, which was not the >>>> intention. >>>> >>>> For now I've sidestepped the audible problem by playing ${greeting} in >>>> another extension that does execute_extension on this one. But the >>>> greeting is still getting set multiple times per call. >>>> >>>> Would appreciate any clues as to why an inner action gets executed even >>>> though the outer condition is false. >>>> >>>> Thanks! >>>> Frank >>>> >>>> >>>> >>>> >>> expression="^set_greeting_tod$"> >>>> >>>> >>>> >>> data="greeting=ivr/ivr-hello.wav"/> >>>> >>>> >>>> >>>> >>> data="greeting=ivr/ivr-good_morning.wav"/> >>>> >>>> >>>> >>>> >>> data="greeting=ivr/ivr-hello.wav"/> >>>> >>>> >>>> >>>> >>> data="greeting=ivr/ivr-good_afternoon.wav"/> >>>> >>>> >>>> >>> break="on-true"> >>>> >>> data="greeting=ivr/ivr-good_evening.wav"/> >>>> >>>> >>>> >>>> >>> data="greeting=ivr/ivr-hello.wav"/> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ing.antonyam at gmail.com Wed Nov 12 21:20:17 2014 From: ing.antonyam at gmail.com (Antony Aguirre Morales) Date: Wed, 12 Nov 2014 12:20:17 -0600 Subject: [Freeswitch-users] [freeswitch-users] configure one extencion in many sofphone and ringing in all Message-ID: Hi. I want to set one extention in 2 different softphone and I have been successful, the problem arises when I call the extention only sounds in one of 2 softphone configured. How could make sounds in the 2 softphone simultaneously. Thnks. -- Ing. Antony Aguirre Morales Ingeniero en Sistemas Computacionales. Celular. 8181729669 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/8d690708/attachment-0001.html From bordmi at rarus.ru Wed Nov 12 23:16:29 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Thu, 13 Nov 2014 00:16:29 +0400 Subject: [Freeswitch-users] [freeswitch-users] configure one extencion in many sofphone and ringing in all In-Reply-To: References: Message-ID: If you want two softphone ringing, it is no special configuration except +A or +E sufix to dialstring. But if you want to talk simultaneously, you must configure conference. There is one variant in vanilla configuration where you can add third party to call in listen only mode (it is known as eavesdropping). All variants of configuration situated in vanilla and described at freeswitch wiki. 12.11.2014 21:30 ???????????? "Antony Aguirre Morales" < ing.antonyam at gmail.com> ???????: > Hi. > > I want to set one extention in 2 different softphone and I have been > successful, the problem arises when I call the extention only sounds in one > of 2 softphone configured. > How could make sounds in the 2 softphone simultaneously. > > Thnks. > > -- > Ing. Antony Aguirre Morales > Ingeniero en Sistemas Computacionales. > Celular. 8181729669 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/91aabf52/attachment.html From brian at freeswitch.org Thu Nov 13 00:45:39 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 12 Nov 2014 15:45:39 -0600 Subject: [Freeswitch-users] Music on hold sample rate mismatch In-Reply-To: References: Message-ID: Its a debug line, not a warning. src/switch_core_file.c: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "File %s sample rate %d doesn't match requested rate %d\n", file_path, fh->samplerate, rate); You can ignore it, its informational only. On Wed, Nov 12, 2014 at 11:27 AM, Oleg Stolyar wrote: > This might be the easiest question ever asked on this list. > > I am playing wav files with sample rate 48000 as music on hold for a > conference with sample rate 8000. I am getting this warning in the logs: > > switch_core_file.c:211 File > /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't > match requested rate 8000 > > I can ignore it, right? This will not create any actual problems? I > don't think it will even increase the CPU consumption but wanted to confirm. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/71c38876/attachment.html From djdevil8 at gmail.com Thu Nov 13 01:19:04 2014 From: djdevil8 at gmail.com (Djdevil) Date: Wed, 12 Nov 2014 16:19:04 -0600 Subject: [Freeswitch-users] [freeswitch-users] configure one extencion in many sofphone and ringing in all In-Reply-To: References: Message-ID: <1FD488B8-0C78-4A2B-8736-DA62E8D33F68@gmail.com> Hi, Thanks a lot by your answer. My question was related when i have an extensi?n and i configure the same extensi?n in two softphones. When this extensi?n is reciving a call, one extensi?n is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. Enviado desde mi iPhone > El 12/11/2014, a las 14:16, ???????, ??????? escribi?: > > If you want two softphone ringing, it is no special configuration except +A or +E sufix to dialstring. But if you want to talk simultaneously, you must configure conference. There is one variant in vanilla configuration where you can add third party to call in listen only mode (it is known as eavesdropping). All variants of configuration situated in vanilla and described at freeswitch wiki. > > 12.11.2014 21:30 ???????????? "Antony Aguirre Morales" ???????: >> Hi. >> >> I want to set one extention in 2 different softphone and I have been successful, the problem arises when I call the extention only sounds in one of 2 softphone configured. >> How could make sounds in the 2 softphone simultaneously. >> >> Thnks. >> >> -- >> Ing. Antony Aguirre Morales >> Ingeniero en Sistemas Computacionales. >> Celular. 8181729669 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/8fa4673f/attachment.html From mike at jerris.com Thu Nov 13 00:51:59 2014 From: mike at jerris.com (Michael Jerris) Date: Wed, 12 Nov 2014 16:51:59 -0500 Subject: [Freeswitch-users] Music on hold sample rate mismatch In-Reply-To: References: Message-ID: Although its useful information. It will work fine, but requires more cpu resources than if you were playing files that already matched the rate. > On Nov 12, 2014, at 4:45 PM, Brian West wrote: > > Its a debug line, not a warning. > > src/switch_core_file.c: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "File %s sample rate %d doesn't match requested rate %d\n", file_path, fh->samplerate, rate); > > You can ignore it, its informational only. > > > On Wed, Nov 12, 2014 at 11:27 AM, Oleg Stolyar > wrote: > This might be the easiest question ever asked on this list. > > I am playing wav files with sample rate 48000 as music on hold for a conference with sample rate 8000. I am getting this warning in the logs: > > switch_core_file.c:211 File /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't match requested rate 8000 > > I can ignore it, right? This will not create any actual problems? I don't think it will even increase the CPU consumption but wanted to confirm. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/d8e1085c/attachment-0001.html From olegstolyar at gmail.com Thu Nov 13 00:53:12 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Nov 2014 13:53:12 -0800 Subject: [Freeswitch-users] Music on hold sample rate mismatch In-Reply-To: References: Message-ID: Thanks Brian. On Wed, Nov 12, 2014 at 1:45 PM, Brian West wrote: > Its a debug line, not a warning. > > src/switch_core_file.c: switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_DEBUG, "File %s sample rate %d doesn't match requested rate > %d\n", file_path, fh->samplerate, rate); > > You can ignore it, its informational only. > > On Wed, Nov 12, 2014 at 11:27 AM, Oleg Stolyar > wrote: > >> This might be the easiest question ever asked on this list. >> >> I am playing wav files with sample rate 48000 as music on hold for a >> conference with sample rate 8000. I am getting this warning in the logs: >> >> switch_core_file.c:211 File >> /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't >> match requested rate 8000 >> >> I can ignore it, right? This will not create any actual problems? I >> don't think it will even increase the CPU consumption but wanted to confirm. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/3aff2f48/attachment.html From ing.antonyam at gmail.com Thu Nov 13 01:04:55 2014 From: ing.antonyam at gmail.com (Antony Aguirre Morales) Date: Wed, 12 Nov 2014 16:04:55 -0600 Subject: [Freeswitch-users] Configure an extencion in both devices and ringing in all. Message-ID: Hi, Thanks a lot by your answer. My question was related when i have an extensi?n and i configure the same extensi?n in two softphones. When this extensi?n is reciving a call, one extensi?n is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/6a3d8419/attachment.html From olegstolyar at gmail.com Thu Nov 13 01:28:46 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Wed, 12 Nov 2014 14:28:46 -0800 Subject: [Freeswitch-users] Music on hold sample rate mismatch In-Reply-To: References: Message-ID: Thanks Michael! Can I assume that the CPU resource usage will still be trivial since it's a shared stream for all the people on hold? On Wed, Nov 12, 2014 at 1:51 PM, Michael Jerris wrote: > Although its useful information. It will work fine, but requires more cpu > resources than if you were playing files that already matched the rate. > > On Nov 12, 2014, at 4:45 PM, Brian West wrote: > > Its a debug line, not a warning. > > src/switch_core_file.c: switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_DEBUG, "File %s sample rate %d doesn't match requested rate > %d\n", file_path, fh->samplerate, rate); > > You can ignore it, its informational only. > > On Wed, Nov 12, 2014 at 11:27 AM, Oleg Stolyar > wrote: > >> This might be the easiest question ever asked on this list. >> >> I am playing wav files with sample rate 48000 as music on hold for a >> conference with sample rate 8000. I am getting this warning in the logs: >> >> switch_core_file.c:211 File >> /usr/local/freeswitch/sounds/music/8000/music.wav sample rate 48000 doesn't >> match requested rate 8000 >> >> I can ignore it, right? This will not create any actual problems? I >> don't think it will even increase the CPU consumption but wanted to confirm. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/d382e4f6/attachment.html From ing.antonyam at gmail.com Thu Nov 13 02:19:59 2014 From: ing.antonyam at gmail.com (antonyam) Date: Wed, 12 Nov 2014 16:19:59 -0700 (MST) Subject: [Freeswitch-users] ringing in two devices configurate one extencion. Message-ID: <1415834399534-7596133.post@n2.nabble.com> Hi, My question was related when i have an extensi?n and i configure the same extensi?n in two softphones. When this extensi?n is reciving a call, one extensi?n is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ringing-in-two-devices-configurate-one-extencion-tp7596133.html Sent from the freeswitch-users mailing list archive at Nabble.com. From fvillarroel at yahoo.com Thu Nov 13 03:13:44 2014 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Wed, 12 Nov 2014 16:13:44 -0800 Subject: [Freeswitch-users] ringing in two devices configurate one extencion. In-Reply-To: <1415834399534-7596133.post@n2.nabble.com> References: <1415834399534-7596133.post@n2.nabble.com> Message-ID: <1415837624.6817.YahooMailNeo@web162006.mail.bf1.yahoo.com> http://wiki.freeswitch.org/wiki/Sofia-SIP#Multiple_Registrations On Wednesday, November 12, 2014 8:23 PM, antonyam wrote: Hi, My question was related when i have an extensi?n and i configure the same extensi?n in two softphones. When this extensi?n is reciving a call, one extensi?n is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ringing-in-two-devices-configurate-one-extencion-tp7596133.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/f345924d/attachment-0001.html From msc at freeswitch.org Thu Nov 13 06:08:54 2014 From: msc at freeswitch.org (Michael Collins) Date: Wed, 12 Nov 2014 19:08:54 -0800 Subject: [Freeswitch-users] Curious TLS issue: "tls.pem" file Message-ID: Hello all, I have been attempting to set up a CentOS (yeah, I know...) system for a buddy and the TLS on the internal profile is causing a failure. I did a sofia loglevel tport 9 and then loaded the internal profile. I see a curious reference to /usr/conf/ssl/tls.pem: 2014-11-12 18:51:26.223262 [DEBUG] sofia.c:2747 Creating agent for internal tport.c:498 tport_tcreate() tport_create(): 0x7f0cf8046840 tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to */EXTERN_IP_ADDR:5060/sip tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): calling tport_listen for udp tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new primary tport 0x7f0cf8021be0 tport.c:751 tport_listen() tport_listen(0x7f0cf8021be0): listening at udp/EXTERN_IP_ADDR:5060/sip tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): calling tport_listen for tcp tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new primary tport 0x7f0cf8070d30 tport.c:751 tport_listen() tport_listen(0x7f0cf8070d30): listening at tcp/EXTERN_IP_ADDR:5060/sip tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to tls/EXTERN_IP_ADDR:5061/sips tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): calling tport_listen for tls tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new primary tport 0x7f0cf8066c90 tport_type_tls.c:239 tport_tls_init_master() tport_tls_init_master(0x7f0cf8066c90): tls key = /usr/conf/ssl/tls.pem tport_tls.c:353 tls_init_context() tls_init_context: invalid local certificate: /usr/conf/ssl/tls.pem tport_tls.c:158 tls_log_errors() tls_init_context: 0200100d:system library:fopen:Permission denied tport_tls.c:158 tls_log_errors() tls_init_context: 20074002:BIO routines:FILE_CTRL:system lib tport_tls.c:158 tls_log_errors() tls_init_context: 140ad002:SSL routines:SSL_CTX_use_certificate_file:system lib tport_tls.c:367 tls_init_context() tls_init_context: invalid private key: /usr/conf/ssl/tls.pem tport_tls.c:158 tls_log_errors() tls_init_context(key): 0200100d:system library:fopen:Permission denied tport_tls.c:158 tls_log_errors() tls_init_context(key): 20074002:BIO routines:FILE_CTRL:system lib tport_tls.c:158 tls_log_errors() tls_init_context(key): 140b0002:SSL routines:SSL_CTX_use_PrivateKey_file:system lib tport_tls.c:379 tls_init_context() tls_init_context: private key does not match the certificate public key tport_tls.c:391 tls_init_context() tls_init_context: error loading CA list: cafile.pem tport_tls.c:158 tls_log_errors() tls_init_context(CA): 140a80b1:SSL routines:SSL_CTX_check_private_key:no certificate assigned tport_tls.c:158 tls_log_errors() tls_init_context(CA): 02001002:system library:fopen:No such file or directory tport_tls.c:158 tls_log_errors() tls_init_context(CA): 2006d080:BIO routines:BIO_new_file:no such file tport_tls.c:158 tls_log_errors() tls_init_context(CA): 0b084002:x509 certificate routines:X509_load_cert_crl_file:system lib tport.c:727 tport_listen() tport_listen(0x7f0cf8046840): tls_init_master(pf=2 tls/[EXTERN_IP_ADDR]:5061): Input/output error tport.c:555 tport_destroy() tport_destroy(0x7f0cf8046840) 2014-11-12 18:51:26.223262 [ERR] sofia.c:2847 Error Creating SIP UA for profile: internal (sip:mod_sofia at EXTERN_IP_ADDR:5060;transport=udp,tcp) ATTEMPT 1 (RETRY IN 5 SEC) I can't find any tls.pem file referred to in any config file and a google search of "tls.pem" yields many references to agent.pem, key.pem, foo.pem but never "tls.pem"... The gentls stuff in the wiki all seemed to work as I saw no errors and I got agent.pem and cafile.pem and other miscellaneous files. Any thoughts on this? Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141112/e7b7e59e/attachment.html From nick.zaitsev at mail.ru Thu Nov 13 10:38:39 2014 From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=) Date: Thu, 13 Nov 2014 10:38:39 +0300 Subject: [Freeswitch-users] =?utf-8?q?Skinny_issue=287985=29?= Message-ID: <1415864319.459776373@f120.i.mail.ru> Good day to you. I try to set the Cisco 7985 to work with freeswitch. But i have a problem: 7985 requests the firmware version and there is a string "Configuring IP" on the 7985 display,however i am able to make and recieve calls with this phone. Here is the debug output 2014-11-13 10:31:55.044796 [DEBUG] mod_skinny.c:1735 [_undef_:0 @ 10.4.33.12:23271] Connection Open 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:2453 [_undef_:0 @ 10.4.33.12:23271] Received RegisterMessage (type=1,length=56). 2014-11-13 10:31:55.064823 [DEBUG] mod_skinny.c:1469 [_undef_:0 @ 10.4.33.12:23271] Clean device from DB with name 'SEP0050600361BC' 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:1237 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Register Ack with Keep Alive (60), Date Format (D/M/Y), Secondary Keep Alive (60) 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:1240 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Capabilities Req 2014-11-13 10:31:55.084802 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received PortMessage (type=2,length=8). 2014-11-13 10:31:55.124820 [DEBUG] mod_skinny.c:2469 MWI Event received for account skinny:11006 at 10.160.1.18 with messages waiting no 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received DeviceUpdateCapabilities (type=30,length=1844). 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2412 Codecs PCMU,PCMA,G729,G729,G729,G729,G722,G722,G722 supported. 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received VersionReqMessage (type=f,length=4). 2014-11-13 10:31:55.144795 [WARNING] skinny_server.c:1762 Device SEP0050600361BC:1 is requesting for firmware version, but none is set. 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:1767 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Version with Version() 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received HeadsetStatusMessage (type=2b,length=8). 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2159 [SEP0050600361BC:1 @ 10.4.33.12:23271] Update headset accessory status (OnHook) 2014-11-13 10:31:55.164801 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received ButtonTemplateReqMessage (type=e,length=4). 2014-11-13 10:31:55.164801 [DEBUG] skinny_server.c:1735 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending ButtonTemplateResMessage (type=97,length=100). 2014-11-13 10:31:55.184802 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received SoftKeyTemplateReqMessage (type=28,length=4). 2014-11-13 10:31:55.184802 [DEBUG] skinny_server.c:2135 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Soft Key Template Request with Default Template 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received SoftKeySetReqMessage (type=25,length=4). 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:1964 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Soft Key Set Request with Set (default) 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:1974 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Select Soft Keys with Line Instance (0), Call ID (0), Soft Key Set (0), Valid Key Mask (ffff) 2014-11-13 10:31:55.224804 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received ConfigStatReqMessage (type=c,length=4). 2014-11-13 10:31:55.224804 [DEBUG] skinny_server.c:1641 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending ConfigStatResMessage (type=93,length=116). 2014-11-13 10:31:55.464803 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received LineStatReqMessage (type=b,length=8). 2014-11-13 10:31:55.464803 [DEBUG] skinny_server.c:1586 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending LineStatResMessage (type=92,length=116). 2014-11-13 10:31:55.724816 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received ForwardStatReqMessage (type=9,length=8). 2014-11-13 10:31:55.724816 [DEBUG] skinny_server.c:1552 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Forward Stat Req Message with Line Instance (1) 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received SpeedDialStatReqMessage (type=a,length=8). 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:1564 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Speed Dial Stat Request for Number (1) 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:1568 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Speed Dial Stat Res with Number (1), Line (11006), Label (Artem Shepelev) 2014-11-13 10:31:55.944813 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received Server Request Message (type=12,length=4). 2014-11-13 10:31:55.944813 [INFO] skinny_server.c:2424 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received Server Request Message (length=4). 2014-11-13 10:31:55.944813 [DEBUG] skinny_server.c:2426 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Server Request Response with IP (10.160.1.21) and Port (2000) 2014-11-13 10:31:55.944813 [DEBUG] skinny_protocol.c:685 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending ServerResponseMessage (type=9e,length=284). 2014-11-13 10:31:55.964799 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received TimeDateReqMessage (type=d,length=4). 2014-11-13 10:31:55.964799 [DEBUG] skinny_protocol.c:850 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Define Time Date with 2014-11-13 10:31:55.964, Timestamp (1415863915), DOW (4) here 10.160.1.21 - freeswitch ip-address in the device config file( SEP0050600361BC.xml)? I've written this for firmware: Could you advise,how I can solve this issue? Thank you for your time and for your help, Best regards. -- Nick Zaitsev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/4b4a9d9e/attachment.html From nick.zaitsev at mail.ru Thu Nov 13 11:48:01 2014 From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=) Date: Thu, 13 Nov 2014 11:48:01 +0300 Subject: [Freeswitch-users] =?utf-8?q?skinny_issue?= Message-ID: <1415868481.187636708@f176.i.mail.ru> Good day to you,again. I have one more skinny issue: I would like to make a group call to skinny phones and sip phones.So because of group_call does not work in case of skinny phones, i've add this to dialplan: but when i answer the call on the 11006(cisco 7985),i have this in debug 2014-11-13 11:43:44.964830 [DEBUG] switch_core_session.c:1614 Session 146 (SKINNY/internal/11006) Locked, Waiting on external entities 2014-11-13 11:43:45.084825 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received OpenReceiveChannelAckMessage (type=22,length=24). 2014-11-13 11:43:45.084825 [WARNING] skinny_server.c:1945 [SEP0050600361BC:1 @ 10.4.33.12:23271] Unable to find session for call id=84. Could you help, please? Thank you for your time, Best regards -- Nick Zaitsev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/79dbd02d/attachment-0001.html From my.post at hotmail.com Thu Nov 13 13:26:49 2014 From: my.post at hotmail.com (Pavel) Date: Thu, 13 Nov 2014 16:26:49 +0600 Subject: [Freeswitch-users] Trying to originate a call using ES. In-Reply-To: References: Message-ID: Hello! Some clarifications: This is working: $con->sendRecv("api originate {originate_timeout=5}user/1000 &park()") But this is not: $con->sendRecv("api originate {return_ring_ready=true,originate_timeout=5}user/1000 &park()") The originated call keeps ringing.Please advise. Thanks. Regards, Pavel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/f0af1d98/attachment.html From vladget at gmail.com Thu Nov 13 13:42:15 2014 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Thu, 13 Nov 2014 12:42:15 +0200 Subject: [Freeswitch-users] failure_causes and DC code... Message-ID: Looks like failure_causes do something wrong... dialplan: At sip: SIP/2.0 480 Temporarily Unavailable. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. with commented at sip 503 with 34 as expected... -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/f931151e/attachment.html From soulofmischief87 at gmail.com Thu Nov 13 11:01:08 2014 From: soulofmischief87 at gmail.com (Tito Cumpen) Date: Thu, 13 Nov 2014 03:01:08 -0500 Subject: [Freeswitch-users] Recording calls. Message-ID: Group, I have a question regarding the recording feature provided in freeswitch. I am currently using freeswitch to proxy h264 and vp8 media and I am uncertain what will happen if I chose to record the call. Will only the audio be recorded ? Is there way method to recording either audio , video or both? Any guidelines on how can the recording be sent with the call id via a rest post? Thanks, Tito -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/1fea9c5d/attachment.html From bordmi at rarus.ru Thu Nov 13 11:23:49 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Thu, 13 Nov 2014 12:23:49 +0400 Subject: [Freeswitch-users] [freeswitch-users] configure one extencion in many sofphone and ringing in all In-Reply-To: <1FD488B8-0C78-4A2B-8736-DA62E8D33F68@gmail.com> References: <1FD488B8-0C78-4A2B-8736-DA62E8D33F68@gmail.com> Message-ID: your entry must view like this: ... 2014-11-13 1:19 GMT+03:00 Djdevil : > Hi, > > Thanks a lot by your answer. > > My question was related when i have an extensi?n and i configure the same > extensi?n in two softphones. When this extensi?n is reciving a call, one > extensi?n is ringing the second one don't. My question is, how can i > configure for ringing both softphones and ringing in both devices? > > Regards. > > Enviado desde mi iPhone > > El 12/11/2014, a las 14:16, ???????, ??????? escribi?: > > If you want two softphone ringing, it is no special configuration except > +A or +E sufix to dialstring. But if you want to talk simultaneously, you > must configure conference. There is one variant in vanilla configuration > where you can add third party to call in listen only mode (it is known as > eavesdropping). All variants of configuration situated in vanilla and > described at freeswitch wiki. > 12.11.2014 21:30 ???????????? "Antony Aguirre Morales" < > ing.antonyam at gmail.com> ???????: > >> Hi. >> >> I want to set one extention in 2 different softphone and I have been >> successful, the problem arises when I call the extention only sounds in one >> of 2 softphone configured. >> How could make sounds in the 2 softphone simultaneously. >> >> Thnks. >> >> -- >> Ing. Antony Aguirre Morales >> Ingeniero en Sistemas Computacionales. >> Celular. 8181729669 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ? ?????????, ??????? ??????? ????????????? ?? ?????????? VoIP ??????? ????????? ????????????? ?????? ???????? ?????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/968db157/attachment.html From mike at jerris.com Thu Nov 13 16:26:00 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Nov 2014 08:26:00 -0500 Subject: [Freeswitch-users] Trying to originate a call using ES. In-Reply-To: References: Message-ID: what return_ring_ready means is to consider the call successful on a sip 180 or ringing indication, at that point, we would stop tracking the timeouts. If you want to wait for the timeout until its answered, don't use return_ring_ready. > On Nov 13, 2014, at 5:26 AM, Pavel wrote: > > > Hello! > Some clarifications: > > This is working: > $con->sendRecv("api originate {originate_timeout=5}user/1000 &park()") > > But this is not: > $con->sendRecv("api originate {return_ring_ready=true,originate_timeout=5}user/1000 &park()") > > > The originated call keeps ringing.Please advise. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/a46352a8/attachment-0001.html From mike at jerris.com Thu Nov 13 16:27:27 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Nov 2014 08:27:27 -0500 Subject: [Freeswitch-users] Recording calls. In-Reply-To: References: Message-ID: We do not currently have capabilities for record/playback/transcoding video. > On Nov 13, 2014, at 3:01 AM, Tito Cumpen wrote: > > I have a question regarding the recording feature provided in freeswitch. I am currently using freeswitch to proxy h264 and vp8 media and I am uncertain what will happen if I chose to record the call. Will only the audio be recorded ? Is there way method to recording either audio , video or both? Any guidelines on how can the recording be sent with the call id via a rest post? > From andrew at cassidywebservices.co.uk Thu Nov 13 16:31:06 2014 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 13 Nov 2014 13:31:06 +0000 Subject: [Freeswitch-users] ringing in two devices configurate one extencion. In-Reply-To: <1415834399534-7596133.post@n2.nabble.com> References: <1415834399534-7596133.post@n2.nabble.com> Message-ID: Enable Multiple Extensions: http://wiki.freeswitch.org/wiki/Sofia-SIP#Multiple_Registrations On 12 November 2014 23:19, antonyam wrote: > Hi, > > My question was related when i have an extensi?n and i configure the same > extensi?n in two softphones. When this extensi?n is reciving a call, one > extensi?n is ringing the second one don't. My question is, how can i > configure for ringing both softphones and ringing in both devices? > > Regards. > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/ringing-in-two-devices-configurate-one-extencion-tp7596133.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/08826300/attachment.html From brian at freeswitch.org Thu Nov 13 16:43:15 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 13 Nov 2014 07:43:15 -0600 Subject: [Freeswitch-users] Curious TLS issue: "tls.pem" file In-Reply-To: References: Message-ID: Its what it will look at if nothing is defined, what exactly have you setup so far for TLS? On Wed, Nov 12, 2014 at 9:08 PM, Michael Collins wrote: > Hello all, > > I have been attempting to set up a CentOS (yeah, I know...) system for a > buddy and the TLS on the internal profile is causing a failure. I did a > sofia loglevel tport 9 and then loaded the internal profile. I see a > curious reference to /usr/conf/ssl/tls.pem: > > 2014-11-12 18:51:26.223262 [DEBUG] sofia.c:2747 Creating agent for internal > tport.c:498 tport_tcreate() tport_create(): 0x7f0cf8046840 > tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to > */EXTERN_IP_ADDR:5060/sip > tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): > calling tport_listen for udp > tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new > primary tport 0x7f0cf8021be0 > tport.c:751 tport_listen() tport_listen(0x7f0cf8021be0): listening at > udp/EXTERN_IP_ADDR:5060/sip > tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): > calling tport_listen for tcp > tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new > primary tport 0x7f0cf8070d30 > tport.c:751 tport_listen() tport_listen(0x7f0cf8070d30): listening at > tcp/EXTERN_IP_ADDR:5060/sip > tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to > tls/EXTERN_IP_ADDR:5061/sips > tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): > calling tport_listen for tls > tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): new > primary tport 0x7f0cf8066c90 > tport_type_tls.c:239 tport_tls_init_master() > tport_tls_init_master(0x7f0cf8066c90): tls key = /usr/conf/ssl/tls.pem > tport_tls.c:353 tls_init_context() tls_init_context: invalid local > certificate: /usr/conf/ssl/tls.pem > tport_tls.c:158 tls_log_errors() tls_init_context: 0200100d:system > library:fopen:Permission denied > tport_tls.c:158 tls_log_errors() tls_init_context: 20074002:BIO > routines:FILE_CTRL:system lib > tport_tls.c:158 tls_log_errors() tls_init_context: 140ad002:SSL > routines:SSL_CTX_use_certificate_file:system lib > tport_tls.c:367 tls_init_context() tls_init_context: invalid private key: > /usr/conf/ssl/tls.pem > tport_tls.c:158 tls_log_errors() tls_init_context(key): 0200100d:system > library:fopen:Permission denied > tport_tls.c:158 tls_log_errors() tls_init_context(key): 20074002:BIO > routines:FILE_CTRL:system lib > tport_tls.c:158 tls_log_errors() tls_init_context(key): 140b0002:SSL > routines:SSL_CTX_use_PrivateKey_file:system lib > tport_tls.c:379 tls_init_context() tls_init_context: private key does not > match the certificate public key > tport_tls.c:391 tls_init_context() tls_init_context: error loading CA > list: cafile.pem > tport_tls.c:158 tls_log_errors() tls_init_context(CA): 140a80b1:SSL > routines:SSL_CTX_check_private_key:no certificate assigned > tport_tls.c:158 tls_log_errors() tls_init_context(CA): 02001002:system > library:fopen:No such file or directory > tport_tls.c:158 tls_log_errors() tls_init_context(CA): 2006d080:BIO > routines:BIO_new_file:no such file > tport_tls.c:158 tls_log_errors() tls_init_context(CA): 0b084002:x509 > certificate routines:X509_load_cert_crl_file:system lib > tport.c:727 tport_listen() tport_listen(0x7f0cf8046840): > tls_init_master(pf=2 tls/[EXTERN_IP_ADDR]:5061): Input/output error > tport.c:555 tport_destroy() tport_destroy(0x7f0cf8046840) > 2014-11-12 18:51:26.223262 [ERR] sofia.c:2847 Error Creating SIP UA for > profile: internal (sip:mod_sofia at EXTERN_IP_ADDR:5060;transport=udp,tcp) > ATTEMPT 1 (RETRY IN 5 SEC) > > I can't find any tls.pem file referred to in any config file and a google > search of "tls.pem" yields many references to agent.pem, key.pem, foo.pem > but never "tls.pem"... > > The gentls stuff in the wiki all seemed to work as I saw no errors and I > got agent.pem and cafile.pem and other miscellaneous files. Any thoughts on > this? > > Thanks! > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/31d74a59/attachment.html From brian at freeswitch.org Thu Nov 13 16:50:35 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 13 Nov 2014 07:50:35 -0600 Subject: [Freeswitch-users] failure_causes and DC code... In-Reply-To: References: Message-ID: Here is some data that may help you understand the cause code mappings in SIP: map QSIG cause codes to SIP from RFC4497 section 8.4.1 see mod_sofia.c: hangup_cause_to_sip map sip responses to QSIG cause codes ala RFC4497 section 8.4.4 see sofia_glue.c: sofia_glue_sip_cause_to_freeswitch You would also need to look at the logs, you've provided no context. On Thu, Nov 13, 2014 at 4:42 AM, Vladimir Getmanshchuk wrote: > Looks like failure_causes do something wrong... > > dialplan: > > > > > > > > > > data="error/normal_circuit_congestion"/> > > > > > > > > > > > At sip: > > SIP/2.0 480 Temporarily Unavailable. > > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > > Content-Length: 0. > with commented > > at sip 503 with 34 as expected... > > -- > Yours sincerely, > Vladimir Getmanshchuk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/6018888b/attachment-0001.html From denis at ringme.ru Thu Nov 13 17:05:31 2014 From: denis at ringme.ru (=?UTF-8?B?0JTQtdC90LjRgQ==?=) Date: Thu, 13 Nov 2014 17:05:31 +0300 Subject: [Freeswitch-users] troubles with regex in mod_ivr? In-Reply-To: References: <5460DED7.3000702@ringme.ru> Message-ID: <5464BAAB.601@ringme.ru> On 12.11.2014 08:36, ???????, ??????? wrote: > From the FreeSWITCH Wiki > (https://wiki.freeswitch.org/wiki/IVR_Menu#How_to_route_the_call_if_no_DTMF_is_pressed): > > > > How to route the call if no DTMF is pressed > > First, define the IVR main menu like this: > > Now we are doing just that. But what if we need exit-action in submenu? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/4a868cf3/attachment.html From nneul at mst.edu Thu Nov 13 17:47:17 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 13 Nov 2014 08:47:17 -0600 Subject: [Freeswitch-users] skinny issue In-Reply-To: <1415868481.187636708@f176.i.mail.ru> References: <1415868481.187636708@f176.i.mail.ru> Message-ID: <5464C475.3010701@mst.edu> Can you narrow this down to a simpler case, and describe the symptom? narrow down what specific thing in your config causes the problem to start. Will likely need a network capture of all traffic to that phone and a more complete log to even begin looking at it if it's a phone that hasn't previously been found to work. (i.e. something that is behaving different than existing phones). -- Nathan On 11/13/2014 02:48 AM, Nick Zaitsev wrote: > > Good day to you,again. > I have one more skinny issue: > I would like to make a group call to skinny phones and sip phones.So because of group_call does not work in case of > skinny phones, i've add this to dialplan: > > > > data="${group_call(Uginfo@${domain_name}+F)},skinny/internal/11000,skinny/internal/11006,skinny/internal/11004,skinny/internal/11003"/> > > > > but when i answer the call on the 11006(cisco 7985),i have this in debug > > 2014-11-13 11:43:44.964830 [DEBUG] switch_core_session.c:1614 Session 146 (SKINNY/internal/11006) Locked, Waiting on > external entities > 2014-11-13 11:43:45.084825 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > OpenReceiveChannelAckMessage (type=22,length=24). > 2014-11-13 11:43:45.084825 [WARNING] skinny_server.c:1945 [SEP0050600361BC:1 @ 10.4.33.12:23271] Unable to find session > for call id=84. > > Could you help, please? > Thank you for your time, > Best regards > -- > Nick Zaitsev > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Thu Nov 13 17:54:19 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 13 Nov 2014 08:54:19 -0600 Subject: [Freeswitch-users] Skinny issue(7985) In-Reply-To: <1415864319.459776373@f120.i.mail.ru> References: <1415864319.459776373@f120.i.mail.ru> Message-ID: <5464C61B.2060408@mst.edu> For the firmware warning - you need to put a section in your skinny profile that looks like this: ... and you'll have to get the matching device type id. From my notes - for a 7985 that should be device type id = 302 Will need a more clear description of the symptom for the rest of it after you've made sure the phone is running current firmware. Side note - I think your firmware specified is probably not correct. 7985's firmware should look something like "cmterm_7985.4-1-7-0" -- Nathan On 11/13/2014 01:38 AM, Nick Zaitsev wrote: > > Good day to you. > > I try to set the Cisco 7985 to work with freeswitch. > But i have a problem: > 7985 requests the firmware version and there is a string "Configuring IP" on the 7985 display,however i am able to make > and recieve calls with this phone. > Here is the debug output > > 2014-11-13 10:31:55.044796 [DEBUG] mod_skinny.c:1735 [_undef_:0 @ 10.4.33.12:23271] Connection Open > 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:2453 [_undef_:0 @ 10.4.33.12:23271] Received RegisterMessage > (type=1,length=56). > 2014-11-13 10:31:55.064823 [DEBUG] mod_skinny.c:1469 [_undef_:0 @ 10.4.33.12:23271] Clean device from DB with name > 'SEP0050600361BC' > 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:1237 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Register Ack with > Keep Alive (60), Date Format (D/M/Y), Secondary Keep Alive (60) > 2014-11-13 10:31:55.064823 [DEBUG] skinny_server.c:1240 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Capabilities Req > 2014-11-13 10:31:55.084802 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received PortMessage > (type=2,length=8). > 2014-11-13 10:31:55.124820 [DEBUG] mod_skinny.c:2469 MWI Event received for account skinny:11006 at 10.160.1.18 with > messages waiting no > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > DeviceUpdateCapabilities (type=30,length=1844). > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2412 Codecs PCMU,PCMA,G729,G729,G729,G729,G722,G722,G722 supported. > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > VersionReqMessage (type=f,length=4). > 2014-11-13 10:31:55.144795 [WARNING] skinny_server.c:1762 Device SEP0050600361BC:1 is requesting for firmware version, > but none is set. > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:1767 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Version with Version() > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > HeadsetStatusMessage (type=2b,length=8). > 2014-11-13 10:31:55.144795 [DEBUG] skinny_server.c:2159 [SEP0050600361BC:1 @ 10.4.33.12:23271] Update headset accessory > status (OnHook) > 2014-11-13 10:31:55.164801 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > ButtonTemplateReqMessage (type=e,length=4). > 2014-11-13 10:31:55.164801 [DEBUG] skinny_server.c:1735 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending > ButtonTemplateResMessage (type=97,length=100). > 2014-11-13 10:31:55.184802 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > SoftKeyTemplateReqMessage (type=28,length=4). > 2014-11-13 10:31:55.184802 [DEBUG] skinny_server.c:2135 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Soft Key Template > Request with Default Template > 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > SoftKeySetReqMessage (type=25,length=4). > 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:1964 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Soft Key Set > Request with Set (default) > 2014-11-13 10:31:55.204801 [DEBUG] skinny_server.c:1974 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Select Soft Keys > with Line Instance (0), Call ID (0), Soft Key Set (0), Valid Key Mask (ffff) > 2014-11-13 10:31:55.224804 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > ConfigStatReqMessage (type=c,length=4). > 2014-11-13 10:31:55.224804 [DEBUG] skinny_server.c:1641 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending > ConfigStatResMessage (type=93,length=116). > 2014-11-13 10:31:55.464803 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > LineStatReqMessage (type=b,length=8). > 2014-11-13 10:31:55.464803 [DEBUG] skinny_server.c:1586 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending > LineStatResMessage (type=92,length=116). > 2014-11-13 10:31:55.724816 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > ForwardStatReqMessage (type=9,length=8). > 2014-11-13 10:31:55.724816 [DEBUG] skinny_server.c:1552 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Forward Stat Req > Message with Line Instance (1) > 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > SpeedDialStatReqMessage (type=a,length=8). > 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:1564 [SEP0050600361BC:1 @ 10.4.33.12:23271] Handle Speed Dial Stat > Request for Number (1) > 2014-11-13 10:31:55.924826 [DEBUG] skinny_server.c:1568 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Speed Dial Stat > Res with Number (1), Line (11006), Label (Artem Shepelev) > 2014-11-13 10:31:55.944813 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received Server Request > Message (type=12,length=4). > 2014-11-13 10:31:55.944813 [INFO] skinny_server.c:2424 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received Server Request > Message (length=4). > 2014-11-13 10:31:55.944813 [DEBUG] skinny_server.c:2426 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending Server Request > Response with IP (10.160.1.21) and Port (2000) > 2014-11-13 10:31:55.944813 [DEBUG] skinny_protocol.c:685 [SEP0050600361BC:1 @ 10.4.33.12:23271] Sending > ServerResponseMessage (type=9e,length=284). > 2014-11-13 10:31:55.964799 [DEBUG] skinny_server.c:2453 [SEP0050600361BC:1 @ 10.4.33.12:23271] Received > TimeDateReqMessage (type=d,length=4). > 2014-11-13 10:31:55.964799 [DEBUG] skinny_protocol.c:850 [SEP0050600361BC:1 @ 10.4.33.12:23271] Send Define Time Date > with 2014-11-13 10:31:55.964, Timestamp (1415863915), DOW (4) > > here 10.160.1.21 - freeswitch ip-address > > in the device config file(SEP0050600361BC.xml) I've written this for firmware: > > > > > Could you advise,how I can solve this issue? > > Thank you for your time and for your help, > Best regards. > > > > -- > Nick Zaitsev > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From sertys at gmail.com Thu Nov 13 17:55:11 2014 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 13 Nov 2014 16:55:11 +0200 Subject: [Freeswitch-users] TLS versions and PFS settings Message-ID: What are the options to enable PFS on TLS support, i found the ability added in a changelog from feb,2014, but can't find the corresponding config file params. Also can i specify a list of allowed ciphers? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/5ce83e29/attachment.html From joelewhite at gmail.com Thu Nov 13 18:37:28 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 13 Nov 2014 10:37:28 -0500 Subject: [Freeswitch-users] Updating via GIT In-Reply-To: <743209AA-89D4-426C-BDE6-E09AE0A156A9@jerris.com> References: <743209AA-89D4-426C-BDE6-E09AE0A156A9@jerris.com> Message-ID: After following Ken's instructions and changing the url from stash.freeswitch.org/... to freeswitch.org/stash/... and running the git checkout and git pull, then finally make current all systems I manage are updated. Thank you everyone for your help :) On Thu, Nov 6, 2014 at 1:04 PM, Michael Jerris wrote: > This all looks good. Are you configuring it to a different location and > somehow running a different version than this is building? > > On Nov 6, 2014, at 11:57 AM, Joel White wrote: > > git pull shows this > > remote: Counting objects: 2381, done. > remote: Compressing objects: 100% (929/929), done. > remote: Total 1445 (delta 1211), reused 566 (delta 504) > Receiving objects: 100% (1445/1445), 238.98 KiB, done. > Resolving deltas: 100% (1211/1211), completed with 368 local objects. > From https://stash.freeswitch.org/scm/fs/freeswitch > ae069dc..b942d0f v1.4 -> origin/v1.4 > dba9abc..7cc5209 fs-video -> origin/fs-video > * [new branch] fs-video2 -> origin/fs-video2 > 8258180..b0050f5 master -> origin/master > * [new branch] mod_smpp34 -> origin/mod_smpp34 > 7512761..6a69eae v1.2.stable -> origin/v1.2.stable > From https://stash.freeswitch.org/scm/fs/freeswitch > * [new tag] v1.4.10 -> v1.4.10 > * [new tag] v1.4.11 -> v1.4.11 > * [new tag] v1.4.12 -> v1.4.12 > * [new tag] v1.5.14 -> v1.5.14 > Updating ae069dc..b942d0f > Fast-forward > Freeswitch.2012.sln | 61 + > Makefile.am | 14 +- > build/Makefile.centos5 | 4 +- > build/Makefile.centos6 | 4 +- > build/Makefile.openbsd | 6 +- > build/Makefile.solaris11 | 4 +- > conf/vanilla/autoload_configs/timezones.conf.xml | 1519 > ++++++++++++++++++-- > conf/vanilla/autoload_configs/xml_rpc.conf.xml | 6 + > conf/vanilla/lang/de/vm/sounds.xml | 6 +- > conf/vanilla/lang/en/vm/sounds.xml | 6 +- > conf/vanilla/lang/he/vm/sounds.xml | 6 +- > conf/vanilla/lang/ru/vm/sounds.xml | 6 +- > conf/vanilla/vars.xml | 16 + > configure.ac | 77 +- > debian/bootstrap.sh | 22 +- > debian/control-modules | 1 + > debian/util.sh | 64 +- > html5/verto/demo/img/cc_banner.gif | Bin 0 -> 36820 bytes > html5/verto/demo/img/cc_banner2014.gif | Bin 42775 -> 0 bytes > html5/verto/demo/img/cc_banner2014.jpg | Bin 791929 -> 0 > bytes > html5/verto/demo/index.html | 4 +- > html5/verto/demo/js/verto-min.js | Bin 52404 -> 52560 > bytes > html5/verto/demo/verto.js | 7 +- > html5/verto/js/src/jquery.FSRTC.js | 11 +- > html5/verto/js/src/jquery.verto.js | 2 + > libs/.gitignore | 1 + > libs/esl/Makefile.am | 4 + > libs/esl/fs_cli.c | 5 +- > libs/esl/managed/ManagedEsl.2012.csproj | 2 +- > libs/freetdm/src/ftmod/ftmod_libpri/ftmod_libpri.c | 2 + > libs/freetdm/src/ftmod/ftmod_zt/ftmod_zt.c | 2 +- > libs/libdingaling/src/libdingaling.c | 13 +- > libs/libdingaling/src/libdingaling.h | 1 + > libs/sofia-sip/.update | 2 +- > libs/sofia-sip/libsofia-sip-ua/sdp/sdp_parse.c | 8 +- > libs/sofia-sip/libsofia-sip-ua/su/sofia-sip/su.h | 2 +- > .../libsofia-sip-ua/su/sofia-sip/su_wait.h | 2 +- > libs/sofia-sip/libsofia-sip-ua/su/su.c | 17 +- > libs/sofia-sip/libsofia-sip-ua/su/su_wait.c | 33 +- > libs/sofia-sip/libsofia-sip-ua/tport/ws.c | 12 +- > libs/srtp/libsrtp.2010.vcxproj | 5 + > libs/srtp/libsrtp.2012.vcxproj | 5 + > libs/win32/openssl/libeay32.2010.vcxproj | 28 +- > libs/win32/openssl/ssleay32.2010.vcxproj | 4 +- > scripts/perl/timezone-gen.pl | 9 +- > src/fs_encode.c | 23 +- > src/include/switch_channel.h | 3 +- > src/include/switch_core_media.h | 5 +- > src/include/switch_stfu.h | 3 +- > src/include/switch_types.h | 17 +- > src/include/switch_utils.h | 1 + > src/mod/applications/mod_commands/mod_commands.c | 25 +- > .../applications/mod_conference/mod_conference.c | 35 +- > src/mod/applications/mod_directory/mod_directory.c | 2 +- > src/mod/applications/mod_snom/mod_snom.c | 38 +- > .../mod_valet_parking/mod_valet_parking.c | 2 +- > src/mod/codecs/mod_g729/Makefile.am | 4 +- > src/mod/codecs/mod_opus/Makefile.am | 4 +- > src/mod/codecs/mod_opus/mod_opus.c | 28 +- > src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj | 135 ++ > src/mod/directories/mod_ldap/Makefile.am | 2 +- > src/mod/endpoints/mod_dingaling/mod_dingaling.c | 9 +- > src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp | 34 + > src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj | 135 ++ > src/mod/endpoints/mod_sofia/mod_sofia.c | 87 +- > src/mod/endpoints/mod_sofia/mod_sofia.h | 22 + > src/mod/endpoints/mod_sofia/sofia.c | 318 +++- > src/mod/endpoints/mod_sofia/sofia_glue.c | 26 +- > src/mod/endpoints/mod_sofia/sofia_media.c | 8 +- > src/mod/endpoints/mod_sofia/sofia_presence.c | 22 +- > src/mod/endpoints/mod_sofia/sofia_reg.c | 97 +- > src/mod/endpoints/mod_verto/mcast/mcast.c | 8 + > src/mod/endpoints/mod_verto/mcast/mcast.h | 20 + > src/mod/endpoints/mod_verto/mcast/mcast_cpp.cpp | 2 + > src/mod/endpoints/mod_verto/mod_verto.2012.vcxproj | 27 +- > src/mod/endpoints/mod_verto/mod_verto.c | 67 +- > src/mod/endpoints/mod_verto/mod_verto.h | 18 +- > src/mod/endpoints/mod_verto/ws.c | 16 +- > src/mod/endpoints/mod_verto/ws.h | 10 +- > src/mod/event_handlers/mod_rayo/mod_rayo.c | 75 +- > .../mod_rayo/rayo_record_component.c | 8 +- > src/mod/event_handlers/mod_rayo/srgs.c | 6 +- > src/mod/event_handlers/mod_rayo/test_srgs/main.c | 9 +- > src/mod/formats/mod_shout/mod_shout.c | 7 +- > src/mod/languages/mod_python/mod_python.c | 2 + > src/mod/xml_int/mod_xml_ldap/Makefile.am | 2 +- > src/mod/xml_int/mod_xml_rpc/mod_xml_rpc.c | 16 + > src/switch_channel.c | 66 +- > src/switch_core_io.c | 12 + > src/switch_core_media.c | 746 +++++++--- > src/switch_core_session.c | 2 +- > src/switch_ivr.c | 26 +- > src/switch_ivr_async.c | 105 +- > src/switch_ivr_play_say.c | 27 + > src/switch_pgsql.c | 2 +- > src/switch_rtp.c | 32 +- > src/switch_stfu.c | 246 +++- > src/switch_utils.c | 19 + > support-d/.bashrc | 4 +- > 99 files changed, 3891 insertions(+), 777 deletions(-) > create mode 100644 html5/verto/demo/img/cc_banner.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.gif > delete mode 100644 html5/verto/demo/img/cc_banner2014.jpg > create mode 100644 src/mod/codecs/mod_vp8/mod_vp8.2012.vcxproj > create mode 100644 src/mod/endpoints/mod_rtc/mod_rtc.2012.vcxproj > > > > On Thu, Nov 6, 2014 at 10:11 AM, Steven Ayre wrote: > >> Try a new git clone from the new url >> >> On 6 November 2014 14:07, Joel White wrote: >> >>> I am having an issue with updating via GIT. I am currently on 1.4.9, >>> but would like to upgrade to the latest 1.4.13 . >>> >>> I have done a GIT pull and did not see the merge error that Ken had >>> mentioned. When I run make current, it goes through the motions and looks >>> to be updating the installation. When I connect back to FS and issue the >>> version command, it still states 1.4.9 >>> >>> How do I correct this? >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/bfb3af77/attachment-0001.html From joelewhite at gmail.com Thu Nov 13 18:44:31 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 13 Nov 2014 10:44:31 -0500 Subject: [Freeswitch-users] Caller ID Message-ID: I have several installations of FreeSWITCH. I have managed to get one to dynamically create the user directory from PostgreSQL and it properly sets the outbound caller id. I have another system, very similar in how it was configured. The second installation also dynamically generates the user directory from PostgreSQL as well. I am however having an issue with setting the caller id on this other system. I have went through with a fine tooth comb and am having a hard time locating the discrepancy between the two systems. Has anyone had an issue similar to this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/55ce40d2/attachment.html From balazs.sandor at virtual-call-center.hu Thu Nov 13 19:00:49 2014 From: balazs.sandor at virtual-call-center.hu (=?UTF-8?B?U8OhbmRvciBCYWzDoXpz?=) Date: Thu, 13 Nov 2014 17:00:49 +0100 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: References: <5460DEF6.6070204@gmail.com> <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> Message-ID: I see them on server side 2014-11-13 16:53:42.291333 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 5:1600 2014-11-13 16:53:42.291333 [DEBUG] switch_channel.c:488 RECV DTMF 5:1600 2014-11-13 16:53:42.291333 [DEBUG] switch_ivr_bridge.c:487 Send signal verto.rtc/#9#9#36202868552#manual#sanya_o [BREAK] but i dont see them in client side (JS) ?dv?zlettel: S?ndor Bal?zs Szoftver fejleszt? Virtual Call Center MUNICH | BUDAPEST | WARSAW Telefon: +36 1 999 7400 Web: www.virtual-call-center.hu 2014-11-11 21:56 GMT+01:00 Anthony Minessale : > Look in your console log. They are probably being received and not > connected to anything. > > > On Tue, Nov 11, 2014 at 2:17 AM, S?ndor Bal?zs < > balazs.sandor at virtual-call-center.hu> wrote: > >> I call a mobile phone from browser with mod_verto, and I would like to >> receive dtmfs in the browser (sent from mobile phone) >> Sorry if I was not clear enough. >> >> ?dv?zlettel: >> >> S?ndor Bal?zs >> >> Szoftver fejleszt? >> >> >> >> Virtual Call Center >> >> MUNICH | BUDAPEST >> | WARSAW >> >> >> Telefon: +36 1 999 7400 >> >> Web: www.virtual-call-center.hu >> >> >> >> >> 2014-11-10 22:48 GMT+01:00 Michael Jerris : >> >>> and by that you mean you want to receive on the server, send from the >>> client? That is built in to mod_verto already, and is functional in the >>> demo at webrtc.freeswitch.org >>> >>> On Nov 10, 2014, at 4:42 PM, S?ndor Bal?zs < >>> balazs.sandor at virtual-call-center.hu> wrote: >>> >>> Receive >>> On 10 Nov 2014 17:17, "Igor Olhovskiy" wrote: >>> >>>> You want to receive or send dtmf via mod_verto? >>>> >>>> On 10.11.14 17:04, Bal?zs S?ndor wrote: >>>> >>>> Hi all! >>>> >>>> I would like to receive dtmf with mod_verto. >>>> How can I do that? >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? > > ? http://freeswitch.org/ ? http://cluecon.com/ ? > http://twitter.com/FreeSWITCH > ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ? sip:888 at conference.freeswitch.org ? +19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/4b588e07/attachment.html From mike at jerris.com Thu Nov 13 19:10:02 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Nov 2014 11:10:02 -0500 Subject: [Freeswitch-users] mode_verto recive dtmf In-Reply-To: References: <5460DEF6.6070204@gmail.com> <7F8CCE9D-23F3-446E-9EEE-383FBC2BA12D@jerris.com> Message-ID: did you look in the js console? > On Nov 13, 2014, at 11:00 AM, S?ndor Bal?zs wrote: > > I see them on server side > 2014-11-13 16:53:42.291333 [DEBUG] switch_rtp.c:6045 RTP RECV DTMF 5:1600 > 2014-11-13 16:53:42.291333 [DEBUG] switch_channel.c:488 RECV DTMF 5:1600 > 2014-11-13 16:53:42.291333 [DEBUG] switch_ivr_bridge.c:487 Send signal verto.rtc/#9#9#36202868552#manual#sanya_o [BREAK] > > but i dont see them in client side (JS) > > 2014-11-11 21:56 GMT+01:00 Anthony Minessale >: > Look in your console log. They are probably being received and not connected to anything. > > > On Tue, Nov 11, 2014 at 2:17 AM, S?ndor Bal?zs > wrote: > I call a mobile phone from browser with mod_verto, and I would like to receive dtmfs in the browser (sent from mobile phone) > Sorry if I was not clear enough. > > 2014-11-10 22:48 GMT+01:00 Michael Jerris >: > and by that you mean you want to receive on the server, send from the client? That is built in to mod_verto already, and is functional in the demo at webrtc.freeswitch.org > >> On Nov 10, 2014, at 4:42 PM, S?ndor Bal?zs > wrote: >> Receive >> >> On 10 Nov 2014 17:17, "Igor Olhovskiy" > wrote: >> You want to receive or send dtmf via mod_verto? >> >> On 10.11.14 17:04, Bal?zs S?ndor wrote: >>> Hi all! >>> >>> I would like to receive dtmf with mod_verto. >>> How can I do that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/83d01f94/attachment-0001.html From brian at freeswitch.org Thu Nov 13 19:19:32 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 13 Nov 2014 10:19:32 -0600 Subject: [Freeswitch-users] TLS versions and PFS settings In-Reply-To: References: Message-ID: in a sofia profile: On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov wrote: > What are the options to enable PFS on TLS support, i found the ability > added in a changelog from feb,2014, but can't find the corresponding config > file params. > Also can i specify a list of allowed ciphers? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/03799da9/attachment.html From max at nysolutions.com Thu Nov 13 19:55:51 2014 From: max at nysolutions.com (Moishe Grunstein) Date: Thu, 13 Nov 2014 16:55:51 +0000 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: Did you compare logs? Sip trace? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support at nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg at 01C72F94.9EE45D60] Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joel White Sent: Thursday, November 13, 2014 10:45 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Caller ID I have several installations of FreeSWITCH. I have managed to get one to dynamically create the user directory from PostgreSQL and it properly sets the outbound caller id. I have another system, very similar in how it was configured. The second installation also dynamically generates the user directory from PostgreSQL as well. I am however having an issue with setting the caller id on this other system. I have went through with a fine tooth comb and am having a hard time locating the discrepancy between the two systems. Has anyone had an issue similar to this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/7a6947b8/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 2424 bytes Desc: image001.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/7a6947b8/attachment.jpg From covici at ccs.covici.com Thu Nov 13 20:27:21 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 13 Nov 2014 12:27:21 -0500 Subject: [Freeswitch-users] cannot compile on latest git Message-ID: <7188.1415899641@ccs.covici.com> Hi. I am unable to compile fs on the latest git as of a few minutes ago. I am getting: CC src/libfreeswitch_la-switch_apr.lo In file included from /usr/include/stdlib.h:24:0, from ./src/include/switch.h:76, from src/switch_apr.c:34: /usr/include/features.h:148:3: error: #warning "_BSD_SOURCE and _SVID_SOURCE are deprecated, use _DEFAULT_SOURCE" [-Werror=cpp] # warning "_BSD_SOURCE and _SVID_SOURCE are deprecated, use _DEFAULT_SOURCE" ^ cc1: all warnings being treated as errors Makefile:1617: recipe for target 'src/libfreeswitch_la-switch_apr.lo' failed Should I file a bug or is this something you know about? I am using gentoo linux c compiler is 4.8.3. Thanks in advance for any suggestions. P.S. I did a bootstrap, config, and make. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From soulofmischief87 at gmail.com Thu Nov 13 20:25:53 2014 From: soulofmischief87 at gmail.com (Tito Cumpen) Date: Thu, 13 Nov 2014 12:25:53 -0500 Subject: [Freeswitch-users] Recording calls. In-Reply-To: References: Message-ID: Ok, This means only the audio channels will be recording right ? Can anyone address my question about abstracting the callid when saving or posting the recording ? On Thu, Nov 13, 2014 at 8:27 AM, Michael Jerris wrote: > We do not currently have capabilities for record/playback/transcoding > video. > > > On Nov 13, 2014, at 3:01 AM, Tito Cumpen > wrote: > > > > I have a question regarding the recording feature provided in > freeswitch. I am currently using freeswitch to proxy h264 and vp8 media and > I am uncertain what will happen if I chose to record the call. Will only > the audio be recorded ? Is there way method to recording either audio , > video or both? Any guidelines on how can the recording be sent with the > call id via a rest post? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/ded8c2ce/attachment.html From mike at jerris.com Thu Nov 13 20:40:45 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Nov 2014 12:40:45 -0500 Subject: [Freeswitch-users] cannot compile on latest git In-Reply-To: <7188.1415899641@ccs.covici.com> References: <7188.1415899641@ccs.covici.com> Message-ID: <6C765432-E697-40F5-BB45-9947780D18FE@jerris.com> You should file a bug. You should tell us if this is due to upgrading your compiler, or a change in our code. If its due to change in our code, nailing down the exact patch that broke it will help us resolve the issue. If it is due to compiler updates, its nice to know if its due to upstream compiler changes, or distro specific patches. If its due to a new compiler, we always appreciate patches that aid in detecting this condition, and changing the code with proper ifdefs to support this. Mike > On Nov 13, 2014, at 12:27 PM, covici at ccs.covici.com wrote: > > Hi. I am unable to compile fs on the latest git as of a few minutes > ago. I am getting: > CC src/libfreeswitch_la-switch_apr.lo > In file included from /usr/include/stdlib.h:24:0, > from ./src/include/switch.h:76, > from src/switch_apr.c:34: > /usr/include/features.h:148:3: > error: #warning "_BSD_SOURCE and > _SVID_SOURCE are deprecated, use > _DEFAULT_SOURCE" [-Werror=cpp] > # warning "_BSD_SOURCE and > _SVID_SOURCE are deprecated, use > _DEFAULT_SOURCE" > ^ > cc1: all warnings being treated > as errors > Makefile:1617: recipe for target > 'src/libfreeswitch_la-switch_apr.lo' > failed > > Should I file a bug or is this something you know about? > > I am using gentoo linux c compiler is 4.8.3. > > Thanks in advance for any suggestions. > > P.S. I did a bootstrap, config, and make. > From vladget at gmail.com Thu Nov 13 21:33:06 2014 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Thu, 13 Nov 2014 20:33:06 +0200 Subject: [Freeswitch-users] failure_causes and DC code... In-Reply-To: References: Message-ID: Brian, I'm really appreciated for your suggestions, but I familiar with sip cause mapping and this implementation in FS. So issue was in hangup... (or on error channels, depends on point of view) After I set failure_cause to any value (except one which includes 34) it resets default pattern of failure causes (btw. where is it in the FS code?) which include 34. So failure_cause not contain 34 and pass dialplan executing after bridge to "error/normal_circuit_congestion". After bridge executed, value of $last_bridge_hangup_cause is empty (btw. Why? I guess it should be NORMAL_CIRCUIT_CONGESTION) After was turn application called hangup which was executed with empty argument value ($last_bridge_hangup_cause is empty).. So hangup returned 480 SIP with 16 Q850 (because it is default in case in case of cause is empty) So I've resolved issue with additional condition of $last_bridge_hangup_cause variable. I guess FS should set variable last_bridge_hangup_cause and other hangup cause variables after execution of bridge to error channels like "error/normal_circuit_congestion"... On Thu, Nov 13, 2014 at 3:50 PM, Brian West wrote: > Here is some data that may help you understand the cause code mappings in > SIP: > > map QSIG cause codes to SIP from RFC4497 section 8.4.1 > > see mod_sofia.c: hangup_cause_to_sip > > map sip responses to QSIG cause codes ala RFC4497 section 8.4.4 > > see sofia_glue.c: sofia_glue_sip_cause_to_freeswitch > > You would also need to look at the logs, you've provided no context. > > > > > On Thu, Nov 13, 2014 at 4:42 AM, Vladimir Getmanshchuk > wrote: > >> Looks like failure_causes do something wrong... >> >> dialplan: >> >> >> >> >> >> >> >> >> >> > data="error/normal_circuit_congestion"/> >> >> >> >> >> >> >> >> >> >> >> At sip: >> >> SIP/2.0 480 Temporarily Unavailable. >> >> Reason: Q.850;cause=16;text="NORMAL_CLEARING". >> >> Content-Length: 0. >> with commented >> >> at sip 503 with 34 as expected... >> >> -- >> Yours sincerely, >> Vladimir Getmanshchuk >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/013d1f0e/attachment.html From aqsyounas at gmail.com Thu Nov 13 21:37:37 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 13 Nov 2014 23:37:37 +0500 Subject: [Freeswitch-users] how to change freeswitch listen ip. Message-ID: Hi, I want freeswitch to listen on local ip 127.0.0.1 instead of public ip. How can i do so.? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/086c7878/attachment.html From smrdoshi at gmail.com Thu Nov 13 21:42:34 2014 From: smrdoshi at gmail.com (Samir Doshi) Date: Fri, 14 Nov 2014 00:12:34 +0530 Subject: [Freeswitch-users] how to change freeswitch listen ip. In-Reply-To: References: Message-ID: Hi, You can simply change sip-ip value to 127.0.0.1 in internal sip profile and reload sofia. It should do your job. Thanks, Samir On Fri, Nov 14, 2014 at 12:07 AM, Aqs Younas wrote: > Hi, > I want freeswitch to listen on local ip 127.0.0.1 instead of public ip. > How can i do so.? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/bce55c7d/attachment-0001.html From aqsyounas at gmail.com Thu Nov 13 21:53:38 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Thu, 13 Nov 2014 23:53:38 +0500 Subject: [Freeswitch-users] how to change freeswitch listen ip. In-Reply-To: References: Message-ID: Thank you very much.:) On 13 November 2014 23:42, Samir Doshi wrote: > Hi, > > You can simply change sip-ip value to 127.0.0.1 in internal sip profile > and reload sofia. > It should do your job. > > Thanks, > Samir > > On Fri, Nov 14, 2014 at 12:07 AM, Aqs Younas wrote: > >> Hi, >> I want freeswitch to listen on local ip 127.0.0.1 instead of public ip. >> How can i do so.? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/91410b9a/attachment.html From ssinyagin at gmail.com Thu Nov 13 21:56:52 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Thu, 13 Nov 2014 19:56:52 +0100 Subject: [Freeswitch-users] Caller ID In-Reply-To: References: Message-ID: Probably your ITSP does not allow you to set the caller ID? Did you run a SIP packet trace to see what caller ID is sent out? On Nov 13, 2014 4:45 PM, "Joel White" wrote: > I have several installations of FreeSWITCH. I have managed to get one to > dynamically create the user directory from PostgreSQL and it properly sets > the outbound caller id. I have another system, very similar in how it was > configured. The second installation also dynamically generates the user > directory from PostgreSQL as well. I am however having an issue with > setting the caller id on this other system. I have went through with a > fine tooth comb and am having a hard time locating the discrepancy between > the two systems. > > Has anyone had an issue similar to this? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/f44895fd/attachment.html From blefko5361 at gmail.com Thu Nov 13 22:11:13 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Thu, 13 Nov 2014 13:11:13 -0600 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" Message-ID: I am trying to tweak mod_spandsp in master and deb package the code for use in my application. I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but it consistently gets stuck at "Building sid-amd64 debs.." First off, I'm not sure if this is the correct way to package custom changes, but also why is the debian build stuck at this step? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/3e07b65c/attachment.html From sertys at gmail.com Thu Nov 13 23:15:42 2014 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 13 Nov 2014 22:15:42 +0200 Subject: [Freeswitch-users] TLS versions and PFS settings In-Reply-To: References: Message-ID: Thanks, Brian. Precise as always. And yet how do i enable perfect forward secrecy? 13 ????. 2014 ?. 18:20 ???????????? "Brian West" ???????: > in a sofia profile: > > > > > > > > > > > > > > > > > > > > > > > > > > On Thu, Nov 13, 2014 at 8:55 AM, Daniel Ivanov wrote: > >> What are the options to enable PFS on TLS support, i found the ability >> added in a changelog from feb,2014, but can't find the corresponding config >> file params. >> Also can i specify a list of allowed ciphers? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/a5378fc0/attachment.html From msc at freeswitch.org Thu Nov 13 23:52:54 2014 From: msc at freeswitch.org (Michael Collins) Date: Thu, 13 Nov 2014 12:52:54 -0800 Subject: [Freeswitch-users] Curious TLS issue: "tls.pem" file In-Reply-To: References: Message-ID: Okay, that was the clue I needed. I hadn't updated the location in the vars.xml so it was looking in the wrong place. The right place also hadn't been set w/ the correct perms. Once I did those two items it all magically worked. Many thanks! -MC On Thu, Nov 13, 2014 at 5:43 AM, Brian West wrote: > Its what it will look at if nothing is defined, what exactly have you > setup so far for TLS? > > > On Wed, Nov 12, 2014 at 9:08 PM, Michael Collins > wrote: > >> Hello all, >> >> I have been attempting to set up a CentOS (yeah, I know...) system for a >> buddy and the TLS on the internal profile is causing a failure. I did a >> sofia loglevel tport 9 and then loaded the internal profile. I see a >> curious reference to /usr/conf/ssl/tls.pem: >> >> 2014-11-12 18:51:26.223262 [DEBUG] sofia.c:2747 Creating agent for >> internal >> tport.c:498 tport_tcreate() tport_create(): 0x7f0cf8046840 >> tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to >> */EXTERN_IP_ADDR:5060/sip >> tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): >> calling tport_listen for udp >> tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): >> new primary tport 0x7f0cf8021be0 >> tport.c:751 tport_listen() tport_listen(0x7f0cf8021be0): listening at >> udp/EXTERN_IP_ADDR:5060/sip >> tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): >> calling tport_listen for tcp >> tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): >> new primary tport 0x7f0cf8070d30 >> tport.c:751 tport_listen() tport_listen(0x7f0cf8070d30): listening at >> tcp/EXTERN_IP_ADDR:5060/sip >> tport.c:1615 tport_bind_server() tport_bind_server(0x7f0cf8046840) to >> tls/EXTERN_IP_ADDR:5061/sips >> tport.c:1685 tport_bind_server() tport_bind_server(0x7f0cf8046840): >> calling tport_listen for tls >> tport.c:621 tport_alloc_primary() tport_alloc_primary(0x7f0cf8046840): >> new primary tport 0x7f0cf8066c90 >> tport_type_tls.c:239 tport_tls_init_master() >> tport_tls_init_master(0x7f0cf8066c90): tls key = /usr/conf/ssl/tls.pem >> tport_tls.c:353 tls_init_context() tls_init_context: invalid local >> certificate: /usr/conf/ssl/tls.pem >> tport_tls.c:158 tls_log_errors() tls_init_context: 0200100d:system >> library:fopen:Permission denied >> tport_tls.c:158 tls_log_errors() tls_init_context: 20074002:BIO >> routines:FILE_CTRL:system lib >> tport_tls.c:158 tls_log_errors() tls_init_context: 140ad002:SSL >> routines:SSL_CTX_use_certificate_file:system lib >> tport_tls.c:367 tls_init_context() tls_init_context: invalid private key: >> /usr/conf/ssl/tls.pem >> tport_tls.c:158 tls_log_errors() tls_init_context(key): 0200100d:system >> library:fopen:Permission denied >> tport_tls.c:158 tls_log_errors() tls_init_context(key): 20074002:BIO >> routines:FILE_CTRL:system lib >> tport_tls.c:158 tls_log_errors() tls_init_context(key): 140b0002:SSL >> routines:SSL_CTX_use_PrivateKey_file:system lib >> tport_tls.c:379 tls_init_context() tls_init_context: private key does not >> match the certificate public key >> tport_tls.c:391 tls_init_context() tls_init_context: error loading CA >> list: cafile.pem >> tport_tls.c:158 tls_log_errors() tls_init_context(CA): 140a80b1:SSL >> routines:SSL_CTX_check_private_key:no certificate assigned >> tport_tls.c:158 tls_log_errors() tls_init_context(CA): 02001002:system >> library:fopen:No such file or directory >> tport_tls.c:158 tls_log_errors() tls_init_context(CA): 2006d080:BIO >> routines:BIO_new_file:no such file >> tport_tls.c:158 tls_log_errors() tls_init_context(CA): 0b084002:x509 >> certificate routines:X509_load_cert_crl_file:system lib >> tport.c:727 tport_listen() tport_listen(0x7f0cf8046840): >> tls_init_master(pf=2 tls/[EXTERN_IP_ADDR]:5061): Input/output error >> tport.c:555 tport_destroy() tport_destroy(0x7f0cf8046840) >> 2014-11-12 18:51:26.223262 [ERR] sofia.c:2847 Error Creating SIP UA for >> profile: internal (sip:mod_sofia at EXTERN_IP_ADDR:5060;transport=udp,tcp) >> ATTEMPT 1 (RETRY IN 5 SEC) >> >> I can't find any tls.pem file referred to in any config file and a google >> search of "tls.pem" yields many references to agent.pem, key.pem, foo.pem >> but never "tls.pem"... >> >> The gentls stuff in the wiki all seemed to work as I saw no errors and I >> got agent.pem and cafile.pem and other miscellaneous files. Any thoughts on >> this? >> >> Thanks! >> -MC >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/10b900d5/attachment-0001.html From freeswitch at guylhem.net Thu Nov 13 23:40:09 2014 From: freeswitch at guylhem.net (Charles Devereaux) Date: Thu, 13 Nov 2014 15:40:09 -0500 Subject: [Freeswitch-users] Configuring a HUAWEI EM770W for gsmopen Message-ID: Hello I'm new to freeswitch, and I would like to try and use a HUAWEI EM770W with gsmopen. The module is unlocked, and voice is enabled. Voice dialing has been tested and works : using MobilePartner on a windows XP running on virtualbox, I can dial a number, and hear the audio. However, from linux I just don't understand where the voice flux can be obtained or how to configure gsmopen. From http://wiki.freeswitch.org/wiki/GSMopen#Configuration_File I thought it would be some ttyUSB device, but I saw nothing. Here's what I did: - disconnected the usb device from virtualbox menu (so that it would be properly initialized with the at commands, etc) - started minicom, typed atd number followed by a semicolon (for voice mode) - monitored /dev/ttyUSB[1-5] : only ttyUSB2 showed some activity : ^CONF:1 ^CONN:1,0 ^CEND:1,69,29,16 MODE:3,3 - checked with AT+CPAS that the communication was active +CPAS: 4 - hanged up with AT+CHUP - checked with AT+CPAS that the communication was no longer active +CPAS: 0 AT+CFUN indicates a proper registration, AT+CSQ returns a proper signal, and no PIN has been configured - I just don't understand how to get the audio or how to configure May I please ask for some help? I will be happy to update the wiki explaining how to use a EM770W (they're cheap!) if it can be made to work. (Also, I have a problem compiling on a debian sid - first error being in bootstrap.sh: automake is not found while 1.14.1 is installed and in the $PATH) Here are the current settings : AT&V &C: 1; &D: 2; &F: 0; E: 1; L: 0; M: 0; Q: 0; V: 1; X: 1; Z: 0; S0: 0; S2: 43; S3: 13; S4: 10; S5: 8; S6: 2; S7: 50; S8: 2; S9: 6; S10: 14; S11: 95; S30: 0; S103: 1; S104: 1; +FCLASS: 0; +ICF: 3,3; +IFC: 2,2; +IPR: 115200; +DR: 0; +DS: 0,0,2048,6; +WS46: 12; +CBST: 0,0,1; +CRLP: (61,61,48,6,0),(61,61,48,6,1),(240,240,52,6,2); +CV120: 1,1,1,0,0,0; +CHSN: 0,0,0,0; +CSSN: 1,1; +CREG: 1; +CGREG: 1; +CFUN:; +CSCS: "IRA"; +CSTA: 129; +CR: 0; +CRC: 0; +CMEE: 1; +CGDCONT: (1,"IP","mm) ; +CGDSCONT: ; +CGTFT: ; +CGEQREQ: (1,2,0,0,0,0,2,0,"0E0","0E0",3,0,0),(2,2,0,0,0,) ; +CGEQMIN: ; +CGQREQ: ; +CGQMIN: ; ; +CGEREP: 0,0; +CGCLASS: "A"; +CGSMS: 3; +CSMS: 0; +CMGF: 0; +CSAS: 0; +CRES: 0; +CSCA: "+13123149810",145; +CSMP: ,,0,0; +CSDH: 0; +CSCB: 0,"",""; +FDD: 0; +FAR: 0; +FCL: 0; +FIT: 0,0; +ES: ,,; +ESA: 0,,,,0,0,255,; +CMOD: 0; +CVHU: 1; ; +CPIN: ????????,????????; +CMEC: 0,0,0; +CKPD: 1,1; +CIND: 0,1,1,1,0,0,1,0; +CMER: 0,0,0,0,0; +CGATT: 1; +CGACT: 0; +CPBS: "ON"; +CPMS: "SM","SM","SM"; +CNMI: 2,1,2,2,0; +CMMS: 2; +FTS: 0; +FRS: 0; +FTH: 3; +FRH: 3; +FTM: 96; +FRM: 96; +CCUG: 0,0,0; +COPS: 1,2,""; +CUSD: 0; +CAOC: 1; +CCWA: 1; +CCLK: ""; +CLVL: 4; +CMUT: 0; +CPOL: 0,2,"",0,0,0; +CPLS: 0; +CTZR: 0; +CTZU: 0; +CLIP: 1; +COLP: 0; +CDIP: 0; +CLIR: 0; ^PORTSEL: 0; ^CPIN: ????????,????????; ^ATRECORD: 0; ^FREQLOCK: 13723540,0; ^GLASTERR: 1; ^CVOICE: 0; ^DDSETEX: 0; ^CMSR: 0; ; ^AUTHDATA: 1,0,"",""; ^CRPN: 0,""; ^WPDST: 1; ^WPDOM: 0; ^WPDFR: 65536,1; ^WPQOS: 255,50; ^WNICT: 0; ; Thanks Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/b7cf34bd/attachment.html From freeswitch at guylhem.net Fri Nov 14 02:26:55 2014 From: freeswitch at guylhem.net (Charles Devereaux) Date: Thu, 13 Nov 2014 18:26:55 -0500 Subject: [Freeswitch-users] Configuring a HUAWEI EM770W for gsmopen In-Reply-To: References: Message-ID: Hello I found some answers for my problems: - For the bootstrap.sh problems on a debian/sid : unset PERL5LIB (to avoid automake problems with a bad environment variable) apt-get install libtool-bin - To get the audio stream: For some reason, you can't do that from ttyUSB0. You must be connected on TTYUSB2 @9600 bauds, and then AT^DDSETEX=2 should send that audio stream to ttyUSB1 - It did work, but only once AT+COPS=? show only 1 network (which I'm not a client of) so I guess I just have a weak signal But at least it's going forward and the EM770 might work :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/ca19e40a/attachment.html From steveayre at gmail.com Fri Nov 14 02:29:34 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 13 Nov 2014 23:29:34 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: Can you clarify - is it still running apparently doing nothing, or does it return to the prompt without finishing building? It's normal to see nothing for a long time after that point. The output of the build itself goes to files in the logs/ directory that'll be created. Try logs/sid-amd64. If it returns to the prompt without building look in that file for errors. On 13 November 2014 19:11, Bruce Lefko wrote: > I am trying to tweak mod_spandsp in master and deb package the code for > use in my application. > > I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but > it consistently gets stuck at "Building sid-amd64 debs.." > > First off, I'm not sure if this is the correct way to package custom > changes, but also why is the debian build stuck at this step? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/6589ec0f/attachment.html From ing.antonyam at gmail.com Fri Nov 14 02:39:25 2014 From: ing.antonyam at gmail.com (Antony Aguirre Morales) Date: Thu, 13 Nov 2014 17:39:25 -0600 Subject: [Freeswitch-users] Set 3 freeswitch to share one single directory. Message-ID: Hi. I have an idea to configure 3 freeswitch but I require that the 3 just consult a directory of extensions, this is possible? Or if there is a way to work simultaneously form two freeswitch directory with a database. better explained in an example: Freeswitch 1 ---- | Freeswitch 2 ---- | ----- Directory of extention Freeswitch 3 ---- | -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/ae6ee42e/attachment.html From rtreleaven at bunnykick.ca Fri Nov 14 03:14:56 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Thu, 13 Nov 2014 19:14:56 -0500 Subject: [Freeswitch-users] Set 3 freeswitch to share one single directory. In-Reply-To: References: Message-ID: I think this is what you want https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl On Thu, Nov 13, 2014 at 6:39 PM, Antony Aguirre Morales < ing.antonyam at gmail.com> wrote: > Hi. > > I have an idea to configure 3 freeswitch but I require that the 3 just > consult a directory of extensions, this is possible? > > Or if there is a way to work simultaneously form two freeswitch directory > with a database. better explained in an example: > > Freeswitch 1 ---- | > Freeswitch 2 ---- | ----- Directory of extention > Freeswitch 3 ---- | > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/48cf160a/attachment.html From blefko5361 at gmail.com Fri Nov 14 04:47:23 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Thu, 13 Nov 2014 19:47:23 -0600 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: Yeah, so it looks like in log/sid-amd64 I see the following over and over again: forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow -> Invoking pbuilder forking: pbuilder create --buildplace /var/cache/pbuilder/base-sid-amd64.cow --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 --distribution sid --no-targz --extrapackages cowdancer W: /root/.pbuilderrc does not exist I: Running in no-targz mode I: Distribution is sid. I: Building the build environment I: running debootstrap /usr/sbin/debootstrap I: Retrieving Release E: Failed getting release file http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release E: debootstrap failed W: Aborting with an error pbuilder create failed On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko wrote: > I am trying to tweak mod_spandsp in master and deb package the code for > use in my application. > > I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but > it consistently gets stuck at "Building sid-amd64 debs.." > > First off, I'm not sure if this is the correct way to package custom > changes, but also why is the debian build stuck at this step? > > Thanks! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141113/418ee880/attachment-0001.html From vladget at gmail.com Fri Nov 14 11:21:47 2014 From: vladget at gmail.com (Vladimir Getmanshchuk) Date: Fri, 14 Nov 2014 10:21:47 +0200 Subject: [Freeswitch-users] bridge failover and failure_causes Message-ID: For the scenario when used multiply dialstring: failure_causes which was sed to USER_BUSY, only works when received 486 (USER BUSY) after 18x.... If it received 486 just after 100 Trying, it trying to bridge to the next gateway and ignore failure_causes=USER_BUSY... Is it normal behavior? -- Yours sincerely, Vladimir Getmanshchuk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/f6984441/attachment.html From fvillarroel at yahoo.com Fri Nov 14 15:18:48 2014 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Fri, 14 Nov 2014 04:18:48 -0800 Subject: [Freeswitch-users] ringing in two devices configurate one extencion. In-Reply-To: <1415837624.6817.YahooMailNeo@web162006.mail.bf1.yahoo.com> References: <1415834399534-7596133.post@n2.nabble.com> <1415837624.6817.YahooMailNeo@web162006.mail.bf1.yahoo.com> Message-ID: <1415967528.11961.YahooMailNeo@web162003.mail.bf1.yahoo.com> http://wiki.freeswitch.org/wiki/Sofia-SIP#Multiple_Registrations On Wednesday, November 12, 2014 9:13 PM, FERNANDO VILLARROEL wrote: http://wiki.freeswitch.org/wiki/Sofia-SIP#Multiple_Registrations On Wednesday, November 12, 2014 8:23 PM, antonyam wrote: Hi, My question was related when i have an extensi?n and i configure the same extensi?n in two softphones. When this extensi?n is reciving a call, one extensi?n is ringing the second one don't. My question is, how can i configure for ringing both softphones and ringing in both devices? Regards. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ringing-in-two-devices-configurate-one-extencion-tp7596133.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/f74468ca/attachment.html From manish.talwar at nexxuspg.com Fri Nov 14 09:52:25 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Fri, 14 Nov 2014 06:52:25 +0000 Subject: [Freeswitch-users] SIP trunking with Nexmo In-Reply-To: <1415745838465.55443@nexxuspg.com> References: <1415264969.3904301.187725825.391FF6CA@webmail.messagingengine.com> <1415377103.4117474.188307717.77C6D87B@webmail.messagingengine.com> <1415475715878.91825@nexxuspg.com>, <1415647026.1146908.189312005.4EEC8C03@webmail.messagingengine.com>, <1415745838465.55443@nexxuspg.com> Message-ID: <1415996595345.35883@nexxuspg.com> Hi, I have found that its connecting a outbound call fine but there is no ring tone coming by FreeSwitch via Nexmo even call is ringing on my mobile. Also, when I answered my connected call then there is no voice stream comes and I cant hear anything at both ends. Please suggest me any configuration if any for getting the ring tone while calling a call and getting a voice at both ends after answering a call. Thanks, Regards, Manish Talwar ________________________________ From: Manish Talwar Sent: 11 November 2014 14:43 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi, Thanks a lot, I am able to make a outbound call by these settings. I have set "absolute_codec_string" as "PCMA,PCMU" and remove "nexmo_forwarded_for" from my dialplan. I have tried this outbound call by "FsClient" and "Wizton" application. Call is running fine with "FsClient" but unable to received the call by "Wizton". In Wizton, Its tried to call on my mobile with similar logs in "FS_CLI" as calling from "FsClient" but after few seconds call was hangup with message as "Originate Failed. Cause: ORIGINATOR_CANCEL". I feel it might be some "codec" configuration problem only, Please find the "FsClient" and "Wizton" log file as an attachment. With, "FsClient" log file I am able to receive the call on my mobile (+919818753995). Also, when its calling Outbound call by "FsClient" then there was no ring sound came on "FsClient" but call was coming on mobile. Please suggest me, what I need to do for Ring sound while its calling a mobile and also Is there any other setting required while calling from "Wizton" or any other medium (like mobile, phone etc). Thanks a lot. Regards, Manish Talwar Also, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 11 November 2014 00:47 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, First, no need to set nexmo_forwarded_for for outbound here, if you re-read my previous email you'll see that it was meant to be used for Nexmo DIDs you use to direct origination traffic into FS. As for the error you are getting with this dialplan, you need to remove "@sip.nexmo.com:5080" from your origination string. Hope it helps, Aviv On Fri, Nov 7, 2014, at 11:10 PM, Manish Talwar wrote: Hi, Thanks for your suggestion, I have make these changes and removed the L16 codec from request now. I have set "absolute_codec_string" and "nexmo_forwarded_for" and its not throwing any error message in SIP trace now. But still, I am not able to make a call on my mobile number "1919818753995". Its show message on FreeSwitch log as "[RECOVERY_ON_TIMER_EXPIRE]" and hangup the freeswitch call. Also, there is no log created on Nexmo dashboard for this call's. I am sending my call request to Nexmo from FreeSwitch by dialplan as. Please find the attached SipTrace file now and let me know what I need to update now. In this log, values passed in "From" and "To" attribute as: From: "18188535351" ;tag=D8g4a5NvH4emF To: I feel there might be some wrong data passed in "To" attribute and it might expecting mobile number "19818753995" only instead on SIP value. Please suggest about these setting also. Thanks, Regards, Manish Talwar ________________________________ From: Aviv Shaham Sent: 07 November 2014 21:48 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo doesn't seem to handle it well if your first specified codec is L16. Try to set absolute_codec_string to PCMU and see if that helps. Also note that there is no need to include custom SIP headers such as api_key, api_secret, and answer_url when you make an outbound call. Since you mentioned also needing inbound - keep in mind that when you use Nexmo's built-in "Forward to SIP" setting for each number in the dashboard, the dialed number will not be passed as a SIP variable and you have no way of knowing it once you receive the SIP invite. One way to get around this is to have your application buy & update numbers via the Nexmo API and set a custom SIP address per Nexmo DID, for example: nexmo_12121115555 at your-server.com and then have a dialplan such as: The nexmo_forwarded_for session variable will now expose to you the dialed Nexmo phone number allowing your application or XML dialplan to use it. Let me know if you are having any other issues. Aviv On Fri, Nov 7, 2014, at 01:05 AM, Manish Talwar wrote: Hi, Thanks for your suggestion, I have tried it and I am able to do a Inbound call via Nexmo now. But still I am not able to make any outbound call from my application. I have checked the FreeSwitch log by siptrace enable and found that my call was terminated with a SIP message as " IP/2.0 407 Proxy Authentication Required". Please find the siptrace log for my call as an attachment. and let me know what changes or configuration I need to make for Proxy Authentication Header. Thanks, Regards, Manish Talwar ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Aviv Shaham Sent: 06 November 2014 14:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP trunking with Nexmo Hi Manish, Nexmo expects your API KEY to be in the From header. To set the caller ID you will need to use "caller-id-name". Good timing btw, I just posted a reply to a similar question on Quora. Have a look: http://qr.ae/DEbk2 - also covers Plivo. Aviv On Thu, Nov 6, 2014, at 12:07 AM, Manish Talwar wrote: Hi, I have make a SIP Trunking (gateway) in FreeSwitch for connecting Nexmo via bridge. I have added this Nexmo file under "\FreeSWITCH\conf\sip_profiles\external" folder. Its successfully registering "sip.nexmo.com" Gateway as mentioned below: Name Type Data State ================================================================================================ external-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5080 RUNNING (0) 192.168.1.140 alias internal ALIASED external profile sip:mod_sofia at 192.168.1.140:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::sip.nexmo.com gateway sip:b9c280dd:7678b8c4 at sip.nexmo.com REGED internal-ipv6 profile sip:mod_sofia@[2001:0:9d38:90d7:102f:3fc4:3f57:fe73]:5060 RUNNING (0) internal profile sip:mod_sofia at 192.168.1.140:5060 RUNNING (0) ================================================================================================ 4 profiles 1 alias But when I send the request to FreeSwitch by Dial command as: 919818753995 here, 18188535351 = Nexmo virtual number for connecting call. 919818753995 = mobile number where I am looking for making a call. It will not connected to Nexmo and call will be terminated with message as: 2014-11-06 11:05:18.088340 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NORMAL_UNSPECIFIED Please find the FreeSwitch call Log and Nexmo Gateway (which I have added in freeswitch conf external folder) as an attachment. Please let me know whether I am doing SIP trunking in correct way or need to change something. Also, Please suggest me what will be my next step for making a call on mobile by this ways. Thanks, Regards, Manish Talwar _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 2 attachments: * FsCall.txt 15k (text/plain) * Nexmo.xml 3k (text/xml) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 9k (text/plain) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Email had 1 attachment: * SipTrace.txt 16k (text/plain) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/61aa4c9b/attachment-0001.html From jobindcruz at gmail.com Fri Nov 14 14:23:22 2014 From: jobindcruz at gmail.com (jobin dcruz) Date: Fri, 14 Nov 2014 16:53:22 +0530 Subject: [Freeswitch-users] How to get bridge call duration with lua Message-ID: Hi, I am using freeswitch bridge,every thing working with fine. My problem is How to get bridge call duration with lua? Cannot update CDR table with bridge call details.How to update it? -- Jobin D'cruz (LAMP Developer) Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/3b34e759/attachment.html From steveayre at gmail.com Fri Nov 14 15:43:47 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 14 Nov 2014 12:43:47 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: FreeSWITCH's build script picks a mirror from your sources.list, you're building on Ubuntu but targetting sid which is not an Ubuntu distribution. You can build if you specify an ubuntu distribution name, but you'll have to provide your own debian/modules.conf file as the build script doesn't have a list of which are compatible with your Ubuntu distributions. On 14 November 2014 01:47, Bruce Lefko wrote: > Yeah, so it looks like in log/sid-amd64 I see the following over and over > again: > > forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow > -> Invoking pbuilder > forking: pbuilder create --buildplace > /var/cache/pbuilder/base-sid-amd64.cow --mirror > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 > --distribution sid --no-targz --extrapackages cowdancer > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Distribution is sid. > I: Building the build environment > I: running debootstrap > /usr/sbin/debootstrap > I: Retrieving Release > E: Failed getting release file > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release > E: debootstrap failed > W: Aborting with an error > pbuilder create failed > > > > On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko wrote: > >> I am trying to tweak mod_spandsp in master and deb package the code for >> use in my application. >> >> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >> but it consistently gets stuck at "Building sid-amd64 debs.." >> >> First off, I'm not sure if this is the correct way to package custom >> changes, but also why is the debian build stuck at this step? >> >> Thanks! >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/52806829/attachment.html From aqsyounas at gmail.com Fri Nov 14 15:50:21 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 14 Nov 2014 17:50:21 +0500 Subject: [Freeswitch-users] How to disable internal profile in freeswitch Message-ID: Is there anyway to disable internal profile in freeswitch.? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/fee698b4/attachment.html From steveayre at gmail.com Fri Nov 14 15:49:57 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 14 Nov 2014 12:49:57 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: Also run this before the util script: export COMPONENTS="main restricted universe multiverse" export NO_COWDANCER_UPDATE=1 (Works around a bug in pbuilder on Ubuntu) To answer your question about changes, what's the modification? Is it something that should be submitted as a patch (correct method is a Stash pull request)? If it's appropriate that'd be a better option... not only does it help the community but it's also easier for you to maintain. If mod_spandsp changes you'll have to modify your patch every time. If not perhaps a fork of the module (different name) is more appropriate. It's possible to use libfreeswitch-dev to build a single module, which you can also build as a .deb package: https://freeswitch.org/stash/projects/FS/repos/freeswitch-contrib/browse/Sevet/debian-packages On 14 November 2014 12:43, Steven Ayre wrote: > FreeSWITCH's build script picks a mirror from your sources.list, you're > building on Ubuntu but targetting sid which is not an Ubuntu distribution. > > You can build if you specify an ubuntu distribution name, but you'll have > to provide your own debian/modules.conf file as the build script doesn't > have a list of which are compatible with your Ubuntu distributions. > > On 14 November 2014 01:47, Bruce Lefko wrote: > >> Yeah, so it looks like in log/sid-amd64 I see the following over and over >> again: >> >> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >> -> Invoking pbuilder >> forking: pbuilder create --buildplace >> /var/cache/pbuilder/base-sid-amd64.cow --mirror >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >> --distribution sid --no-targz --extrapackages cowdancer >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Distribution is sid. >> I: Building the build environment >> I: running debootstrap >> /usr/sbin/debootstrap >> I: Retrieving Release >> E: Failed getting release file >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >> E: debootstrap failed >> W: Aborting with an error >> pbuilder create failed >> >> >> >> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >> wrote: >> >>> I am trying to tweak mod_spandsp in master and deb package the code for >>> use in my application. >>> >>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>> but it consistently gets stuck at "Building sid-amd64 debs.." >>> >>> First off, I'm not sure if this is the correct way to package custom >>> changes, but also why is the debian build stuck at this step? >>> >>> Thanks! >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/8a982f8b/attachment.html From sertys at gmail.com Fri Nov 14 15:50:49 2014 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 14 Nov 2014 14:50:49 +0200 Subject: [Freeswitch-users] Configuring a HUAWEI EM770W for gsmopen In-Reply-To: References: Message-ID: You're very observative, but you should have started by looking at the mod_gsmopen code. It uses the audio_device handle from the config file to get the GSM stream as you have noticed. On the given circumstances, i think just properly configuring the device will run it on gsmopen. On Fri, Nov 14, 2014 at 1:26 AM, Charles Devereaux wrote: > Hello > > I found some answers for my problems: > > - For the bootstrap.sh problems on a debian/sid : > unset PERL5LIB (to avoid automake problems with a bad environment variable) > apt-get install libtool-bin > > - To get the audio stream: > > For some reason, you can't do that from ttyUSB0. You must be connected on > TTYUSB2 @9600 bauds, and then AT^DDSETEX=2 should send that audio stream to > ttyUSB1 > > - It did work, but only once > AT+COPS=? show only 1 network (which I'm not a client of) so I guess I > just have a weak signal > > But at least it's going forward and the EM770 might work :-) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/4690d2b0/attachment-0001.html From steveayre at gmail.com Fri Nov 14 15:59:58 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 14 Nov 2014 12:59:58 +0000 Subject: [Freeswitch-users] How to disable internal profile in freeswitch In-Reply-To: References: Message-ID: Delete sip_profiles/internal.xml (applies on restart) At runtime 'sofia profile internal stop' You can configure as many or as few profiles as you like, there's nothing special about the 'internal' 'external' etc profiles, they're just default examples. On 14 November 2014 12:50, Aqs Younas wrote: > Is there anyway to disable internal profile in freeswitch.? > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/7364d845/attachment.html From aqsyounas at gmail.com Fri Nov 14 16:13:31 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 14 Nov 2014 18:13:31 +0500 Subject: [Freeswitch-users] How to disable internal profile in freeswitch In-Reply-To: References: Message-ID: Thank you very much.! On 14 November 2014 17:59, Steven Ayre wrote: > Delete sip_profiles/internal.xml (applies on restart) > > At runtime 'sofia profile internal stop' > > You can configure as many or as few profiles as you like, there's nothing > special about the 'internal' 'external' etc profiles, they're just default > examples. > > On 14 November 2014 12:50, Aqs Younas wrote: > >> Is there anyway to disable internal profile in freeswitch.? >> >> Thanks >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/9807385a/attachment.html From matsumoto at itsherpa.com Fri Nov 14 16:10:01 2014 From: matsumoto at itsherpa.com (=?UTF-8?B?5p2+5pys56WQ5b+X?=) Date: Fri, 14 Nov 2014 22:10:01 +0900 Subject: [Freeswitch-users] about conference play and bgapi In-Reply-To: <882AF05B1A21418D87AAF24C2F0CC49E@gmail.com> References: <882AF05B1A21418D87AAF24C2F0CC49E@gmail.com> Message-ID: it was solved. I was misunderstanding about "Member-ID". $e->getHeader("Member-ID") Thank you Mr Seven. On Tue, Nov 11, 2014 at 3:22 PM, Seven Du wrote: > not sure how to do that in perl, but you should subscribe to the CUSTOM > conference::maintenance event or ALL event and you could get that > > with $e->getHeader("Member-ID") > > > -- > Seven Du > http://about.me/dujinfang > http://www.dujinfang.com > http://www.freeswitch.org.cn > > Sent with Sparrow > > On Friday, November 7, 2014 at 10:02 PM, ???? wrote: > > Thank you Mr.Seven Du > > I got the sound !. it was my mistake. > because of localhost. It should be IP. > $con = new ESL::ESLconnection("localhost", 8021, ClueCon); > > However, I still have problem. > > Let me know how to get 'member_id' below. in perl. > i tried below. > $e->getHeader("Caller-Caller-ID-Name"); > $e->getHeader("Caller-Caller-ID-Number"); > > conference play [|] > > thank you. > > > On Wed, Nov 5, 2014 at 10:52 AM, Seven Du wrote: > > I don?t quite understand you but you should not put SPACE between % and s. > Try to paste the real code and debug logs on pastebin to get better help. > > On Tuesday, November 4, 2014 at 11:51 PM, ???? wrote: > > Dear Stanislav Sinyagin > > Thank you for your help. > > ????????????????????????????? > > ???Perl??? > > ?????????????????????????wav?????????????????? > > ????????????????????? > my $ api_cmd = sprintf ("conference % s play % s % s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > ???$ con-> bgapi ($ api_cmd); ????????????? > > fs_cli ? ?conference ************* play? ????????????????????????? > > ????????????? > > ??????????? > > > > On Tue, Nov 4, 2014 at 9:28 PM, Stanislav Sinyagin > wrote: > > Dear Matsumoto-san, > > I think it will be easier if you write in Japanese, then it will be clear > how we could help. I know a few Japanese-speaking colleagues who may help in > communicating. > > > > > On Mon, Nov 3, 2014 at 12:11 PM, ???? wrote: > > Hello > > I have two issues. > > I am writing in Perl. > > While 2 people are talking in a conference room, the one person want to play > the sound. > > In "Caller-Username", can you get useless. > I have tried the above but, Member: it will not become a *** not found.. > > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav, $ e-> getHeader ("Caller-Username")) ; > > > I want to play the above in the background > It can not play in the next program. > my $ api_cmd = sprintf ("conference% s play% s% s", $ e-> getHeader > ("Conference-Name"), /etc/a.wav); > > $ con-> bgapi ($ api_cmd); > > Best regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveayre at gmail.com Fri Nov 14 17:39:31 2014 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 14 Nov 2014 14:39:31 +0000 Subject: [Freeswitch-users] Set 3 freeswitch to share one single directory. In-Reply-To: References: Message-ID: This will cover the authentication part, but registrations will need a shared ODBC database on the sofia profile. That won't always work however if the client is behind some types of NAT though, as the port mapping can be specific to the ip the client registered too. To work around that you may need to route the call via the FS server that they registered to. On 14 November 2014 00:14, Russell Treleaven wrote: > I think this is what you want > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl > > On Thu, Nov 13, 2014 at 6:39 PM, Antony Aguirre Morales < > ing.antonyam at gmail.com> wrote: > >> Hi. >> >> I have an idea to configure 3 freeswitch but I require that the 3 just >> consult a directory of extensions, this is possible? >> >> Or if there is a way to work simultaneously form two freeswitch directory >> with a database. better explained in an example: >> >> Freeswitch 1 ---- | >> Freeswitch 2 ---- | ----- Directory of extention >> Freeswitch 3 ---- | >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/523d1e54/attachment-0001.html From Tim at Millicorp.com Fri Nov 14 17:50:46 2014 From: Tim at Millicorp.com (Tim Meade) Date: Fri, 14 Nov 2014 14:50:46 +0000 Subject: [Freeswitch-users] XML_Curl and CentOS 7 Message-ID: <804D48104511D4468F0D60DF9D3100353BC51080@MAIL.millicorp.com> Good day all.... Playing around with CentOS 7 here and compile accomplished and everything loads/runs. Issue is with XML_CURL. The mod loads, but it never seems to fire. Exact same conf setup on CentOS 6.5 works great. Any known issues or thoughts on this? Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/7d6282fb/attachment.html From Tim at Millicorp.com Fri Nov 14 18:00:45 2014 From: Tim at Millicorp.com (Tim Meade) Date: Fri, 14 Nov 2014 15:00:45 +0000 Subject: [Freeswitch-users] XML_Curl and CentOS 7 In-Reply-To: <804D48104511D4468F0D60DF9D3100353BC51080@MAIL.millicorp.com> References: <804D48104511D4468F0D60DF9D3100353BC51080@MAIL.millicorp.com> Message-ID: <804D48104511D4468F0D60DF9D3100353BC51608@MAIL.millicorp.com> Just wanted to add I've confirmed this on stable and latest branches. Configuration XML_CURL does fire during startup. Dialplan does not seem to fire at all which is confirmed with ngrep as there are no calls to the xml provider server. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Friday, November 14, 2014 9:51 AM To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org) Subject: [Freeswitch-users] XML_Curl and CentOS 7 Good day all.... Playing around with CentOS 7 here and compile accomplished and everything loads/runs. Issue is with XML_CURL. The mod loads, but it never seems to fire. Exact same conf setup on CentOS 6.5 works great. Any known issues or thoughts on this? Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/780d0235/attachment.html From krice at freeswitch.org Fri Nov 14 18:02:14 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 14 Nov 2014 15:02:14 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54661976655ed_e518a633307607f@ip-10-58-122-39.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/f1474822/attachment.html From mike at jerris.com Fri Nov 14 18:05:45 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 14 Nov 2014 10:05:45 -0500 Subject: [Freeswitch-users] bridge failover and failure_causes In-Reply-To: References: Message-ID: <52323EEB-AD69-461E-9CFD-1619DF945824@jerris.com> sounds like a bug to me, please file a jira. > On Nov 14, 2014, at 3:21 AM, Vladimir Getmanshchuk wrote: > > For the scenario when used multiply dialstring: > > > failure_causes which was sed to USER_BUSY, > only works when received 486 (USER BUSY) after 18x.... > If it received 486 just after 100 Trying, it trying to bridge to the next gateway and ignore failure_causes=USER_BUSY... > > Is it normal behavior? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/5a7c9ac8/attachment.html From lists at kavun.ch Fri Nov 14 19:14:06 2014 From: lists at kavun.ch (Emrah) Date: Fri, 14 Nov 2014 17:14:06 +0100 Subject: [Freeswitch-users] TLS setup failed Message-ID: Hi list, I am able to use FS with SSLv23 with Blink Pro on Mac OS, but that?s about it. I get the following error if I connect with any other device (Bria iOS, Yealink phone, Join Softphone): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) I came across this thread: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-td7587736.html but it doesn?t seem to apply to my scenario. I am using a commercial certificate. My devices connect to a domain which has an SRV record which points to itself on the SSL port. SSL host is an A record and matches the CN in the certificate. Server cA check is even turned off on certain phones. The only error I get on my Yealink phone is this: Nov 14 11:04:03 SIP [524]: SDL <6+info > [000] SSL_connect (read done) Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL ERROR Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL_connect error I would appreciate to know how I could debug this further. Or if you have any clue at what may be going on. Thanks! Emrah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/f1fca33a/attachment.html From italorossib at gmail.com Fri Nov 14 20:06:08 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Fri, 14 Nov 2014 14:06:08 -0300 Subject: [Freeswitch-users] Doc-Sprint Starting Now Message-ID: Hello all, We're starting the FreeSWITCH doc sprint right now! If you want to help please join #freeswitch-docs and we'll be available to give you instructions. -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/3dafa3f5/attachment-0001.html From msc at freeswitch.org Fri Nov 14 21:15:03 2014 From: msc at freeswitch.org (Michael Collins) Date: Fri, 14 Nov 2014 10:15:03 -0800 Subject: [Freeswitch-users] How to get bridge call duration with lua In-Reply-To: References: Message-ID: Can you describe in a little more detail what you are trying to do? For example, are you trying to get the bridge duration at some random point during the call ("how long has this call been bridged so far?") or after the call has ended? -MC On Fri, Nov 14, 2014 at 3:23 AM, jobin dcruz wrote: > Hi, > > I am using freeswitch bridge,every thing working with fine. > > > data="hangup_after_bridge=true"/> > data="ringback=${us-ring}"/> > data="sofia/gateway/outside/${customer_number}"/> > > My problem is > > How to get bridge call duration with lua? > Cannot update CDR table with bridge call details.How to update it? > > -- > Jobin D'cruz (LAMP Developer) > Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com > > > > Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework > Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI > MooTools Smarty Firefox Extension > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/11cc4dad/attachment.html From blefko5361 at gmail.com Sat Nov 15 00:47:03 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Fri, 14 Nov 2014 15:47:03 -0600 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: So I ran those two exports you gave me and added "-c precise" to my utils call, but it still seems to be running into that pbuilder bug. Here's the output or log/precise-amd64: pbuilder/build//cow.20008 cow-shell W: /root/.pbuilderrc does not exist I: Running in no-targz mode I: Upgrading for distribution precise I: copying local configuration cp: cannot create regular file `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory pbuilder update failed E: could not update with cowdancer, try --no-cowdancer-update option forking: rm -rf /var/cache/pbuilder/build//cow.20008 -> Copying COW directory forking: rm -rf /var/cache/pbuilder/build//cow.22359 forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow /var/cache/pbuilder/build//cow.22359 I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist failed, it didn't exist? -> Invoking pbuilder forking: pbuilder update --override-config --buildplace /var/cache/pbuilder/build//cow.22359 --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise --no-targz --internal-chrootexec chroot /var/cache/pbuilder/build//cow.22359 cow-shell W: /root/.pbuilderrc does not exist I: Running in no-targz mode I: Upgrading for distribution precise I: copying local configuration cp: cannot create regular file `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory pbuilder update failed E: could not update with cowdancer, try --no-cowdancer-update option forking: rm -rf /var/cache/pbuilder/build//cow.22359 cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, does not exist Thanks! On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko wrote: > Yeah, so it looks like in log/sid-amd64 I see the following over and over > again: > > forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow > -> Invoking pbuilder > forking: pbuilder create --buildplace > /var/cache/pbuilder/base-sid-amd64.cow --mirror > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 > --distribution sid --no-targz --extrapackages cowdancer > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Distribution is sid. > I: Building the build environment > I: running debootstrap > /usr/sbin/debootstrap > I: Retrieving Release > E: Failed getting release file > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release > E: debootstrap failed > W: Aborting with an error > pbuilder create failed > > > > On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko wrote: > >> I am trying to tweak mod_spandsp in master and deb package the code for >> use in my application. >> >> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >> but it consistently gets stuck at "Building sid-amd64 debs.." >> >> First off, I'm not sure if this is the correct way to package custom >> changes, but also why is the debian build stuck at this step? >> >> Thanks! >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/516d2da9/attachment.html From freeswitch at guylhem.net Sat Nov 15 01:52:21 2014 From: freeswitch at guylhem.net (Charles Devereaux) Date: Fri, 14 Nov 2014 17:52:21 -0500 Subject: [Freeswitch-users] Configuring a HUAWEI EM770W for gsmopen In-Reply-To: References: Message-ID: Hello On Fri, Nov 14, 2014 at 7:50 AM, Daniel Ivanov wrote: > you should have started by looking at the mod_gsmopen code > Initially I was worried the EM770 would not be compatbile and I wanted to see how and where I could get the audio flux, but yes that would have been more efficient. Newbie mistake, which I'll avoid reiterating by asking for suggestions first. Besides http://wiki.freeswitch.org/wiki/Getting_Started_Guide, is there a tutorial for people who are new to freeswitch, just to get say gsmopen running and expose it for tests? (I would like to run more tests and check if the audio is working, as playing the capture I made from ttyUSB1 with sox only gives me static one 1 of the 3 files, with 2 refusing to play with "error during GSM decode: Numerical result out of range") Based on what I read, I though about using a sip client and sofia to call 9195 (five second delay echo test) would be the best idea, but maybe there are simpler ways. I've read about sipjs for example ( http://sipjs.com/guides/server-configuration/freeswitch/) > On the given circumstances, i think just properly configuring the device > will run it on gsmopen. > Indeed, it is very promising. I got 2 such cards for $15 (about 10 Eur) including delivery, and there was no need for any unlocking. I believe it's the cheapest option to play with gsmopen! Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/2423f800/attachment.html From netcentrica at gmail.com Sat Nov 15 02:07:18 2014 From: netcentrica at gmail.com (Mateusz Bartczak) Date: Sat, 15 Nov 2014 00:07:18 +0100 Subject: [Freeswitch-users] Hack for Zoiper speex codec payload problem In-Reply-To: References: Message-ID: Zoiper sends 101 code, which seems to be not recognized by FS absolute_codec_string is set to speex at 8000h@20i,SILK at 8000h @20i,G729,GSM,SILK at 12000h@20i,PCMA,PCMU Here are parts from logs: 2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: v=0 o=Z 0 0 IN IP4 37.190.147.137 s=Z c=IN IP4 37.190.147.137 t=0 0 m=audio 64037 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv payload to 101 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup sofia/external/ 964938 at 80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION] Still no success 2014-10-21 14:29 GMT+02:00 Brian West : > Since the codec is in the dynamic range the payload number is of little to > no value the codec name is then used to compare. So could you provide some > details of what you think is wrong? > > On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak > wrote: > >> Android, but the same problem is with Windows client >> >> 2014-10-17 12:35 GMT+02:00 Brian West : >> >>> Android or iOS? >>> >>> >>> On Friday, October 17, 2014, Mateusz Bartczak >>> wrote: >>> >>>> Hi All, >>>> >>>> I have problem with getting Zoiper with enabled codec speex/8000 >>>> working with FS. I saw some discussions about this problem before, but none >>>> of them ended with working solution. >>>> >>>> Maybe these days some of you can share some hints how to hack this >>>> >>>> Problem is that Zoiper sends SDP payload type 110 for speex/8000, but >>>> FS expects other type for that codec. >>>> >>>> I tried to rewrite SDP on OpenSIPS proxy that sits between end users >>>> and FS box, and then later rewrite replies with SDP payload generated by FS >>>> to 110 code, but calls always end-up with codec negotiation failure >>>> >>>> And yes, I know that zoiper is broken, please don't paste RFC here :) >>>> >>>> It gets very popular on mobile devices and I'm forced to provide >>>> support for it, even if its broken >>>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141115/e17ec21b/attachment-0001.html From brian at freeswitch.org Sat Nov 15 02:24:14 2014 From: brian at freeswitch.org (Brian West) Date: Fri, 14 Nov 2014 17:24:14 -0600 Subject: [Freeswitch-users] Hack for Zoiper speex codec payload problem In-Reply-To: References: Message-ID: Well considering 101 is telephony-event, Zoiper would appear to be broken. On Fri, Nov 14, 2014 at 5:07 PM, Mateusz Bartczak wrote: > Zoiper sends 101 code, which seems to be not recognized by FS > > absolute_codec_string is set to speex at 8000h@20i,SILK at 8000h > @20i,G729,GSM,SILK at 12000h@20i,PCMA,PCMU > > Here are parts from logs: > > 2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: > v=0 > o=Z 0 0 IN IP4 37.190.147.137 > s=Z > c=IN IP4 37.190.147.137 > t=0 0 > m=audio 64037 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf > send/recv payload to 101 > 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup sofia/external/ > 964938 at 80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > Still no success > > > 2014-10-21 14:29 GMT+02:00 Brian West : > >> Since the codec is in the dynamic range the payload number is of little >> to no value the codec name is then used to compare. So could you provide >> some details of what you think is wrong? >> >> On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak >> wrote: >> >>> Android, but the same problem is with Windows client >>> >>> 2014-10-17 12:35 GMT+02:00 Brian West : >>> >>>> Android or iOS? >>>> >>>> >>>> On Friday, October 17, 2014, Mateusz Bartczak >>>> wrote: >>>> >>>>> Hi All, >>>>> >>>>> I have problem with getting Zoiper with enabled codec speex/8000 >>>>> working with FS. I saw some discussions about this problem before, but none >>>>> of them ended with working solution. >>>>> >>>>> Maybe these days some of you can share some hints how to hack this >>>>> >>>>> Problem is that Zoiper sends SDP payload type 110 for speex/8000, but >>>>> FS expects other type for that codec. >>>>> >>>>> I tried to rewrite SDP on OpenSIPS proxy that sits between end users >>>>> and FS box, and then later rewrite replies with SDP payload generated by FS >>>>> to 110 code, but calls always end-up with codec negotiation failure >>>>> >>>>> And yes, I know that zoiper is broken, please don't paste RFC here :) >>>>> >>>>> It gets very popular on mobile devices and I'm forced to provide >>>>> support for it, even if its broken >>>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/3bcea302/attachment.html From krice at freeswitch.org Sat Nov 15 03:11:24 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 14 Nov 2014 18:11:24 -0600 Subject: [Freeswitch-users] Hack for Zoiper speex codec payload problem In-Reply-To: Message-ID: That?s just broken in general... The RTP map there is saying that codec 101 is telephone-event/8000 which is specifically RFC2833 DTMF... There is no audio codec in there... In this case it appears no actual audio codecs were offered On 11/14/14 5:07 PM, "Mateusz Bartczak" wrote: > Zoiper sends 101 code, which seems to be not recognized by FS > > absolute_codec_string is set to > speex at 8000h@20i,SILK at 8000h@20i,G729,GSM,SILK at 12000h@20i,PCMA,PCMU > > Here are parts from logs: > > 2014-11-15 00:01:26.118326 [DEBUG] sofia.c:5860 Remote SDP: > v=0 > o=Z 0 0 IN IP4 37.190.147.137 > s=Z > c=IN IP4 37.190.147.137 > t=0 0 > m=audio 64037 RTP/AVP 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:32000:20:44000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:16000:20:42200] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SPEEX:99:8000:20:24600] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SILK:118:12000:20:25000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[SILK:117:8000:20:20000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G729:18:8000:20:8000] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5284 Audio Codec Compare > [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200] > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [DEBUG] sofia_glue.c:5453 Set 2833 dtmf send/recv > payload to 101 > 2014-11-15 00:01:26.118326 [NOTICE] sofia.c:6185 Hangup > sofia/external/964938 at 80.72.35.123 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > Still no success > > > 2014-10-21 14:29 GMT+02:00 Brian West : >> Since the codec is in the dynamic range the payload number is of little to no >> value the codec name is then used to compare.? So could you provide some >> details of what you think is wrong? >> >> On Fri, Oct 17, 2014 at 7:34 AM, Mateusz Bartczak >> wrote: >>> Android, but the same problem is with Windows client >>> >>> 2014-10-17 12:35 GMT+02:00 Brian West : >>>> Android or iOS? >>>> >>>> >>>> On Friday, October 17, 2014, Mateusz Bartczak >>>> wrote: >>>>> Hi All, >>>>> >>>>> I have problem with getting Zoiper with enabled codec speex/8000 working >>>>> with FS. I saw some discussions about this problem before, but none of >>>>> them ended with working solution. >>>>> >>>>> Maybe these days some of you can share some hints how to hack this >>>>> >>>>> Problem is that Zoiper sends SDP payload type 110 for speex/8000, but FS >>>>> expects other type for that codec.? >>>>> >>>>> I tried to rewrite SDP on OpenSIPS proxy that sits between end users and >>>>> FS box, and then later rewrite replies with SDP payload generated by FS to >>>>> 110 code, but calls always end-up with codec negotiation failure >>>>> >>>>> And yes, I know that zoiper is broken, please don't paste RFC here :)? >>>>> >>>>> It gets very popular on mobile devices and I'm forced to provide support >>>>> for it, even if its broken >>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141114/d030d281/attachment-0001.html From krice at freeswitch.org Sat Nov 15 04:11:35 2014 From: krice at freeswitch.org (Ken Rice) Date: Sat, 15 Nov 2014 01:11:35 +0000 Subject: [Freeswitch-users] Get Billy Club now on iTunes Message-ID: <5466a847a1744_1c10e53328881b9@ip-10-5-131-236.mail> New Post on freeswitch.org from anthm check it out at http://ift.tt/14orplP Get Billy Club now on iTunes I posted last week about this from my phone but it didn?t work very well. CURSE YOU MOBILE BROWSERS! Anyway, My kid was in a movie 3 years ago when he was 12 and not even old enough to see it but it finally was released and is available on iTunes. He?s a mean baseball player who makes an ordinary kid grow up to be a killer. Check out Billy Club! Only $5 to rent and $10 to Own. http://ift.tt/14orplT -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141115/b7edfd06/attachment.html From jobindcruz at gmail.com Sat Nov 15 05:48:42 2014 From: jobindcruz at gmail.com (jobin dcruz) Date: Sat, 15 Nov 2014 08:18:42 +0530 Subject: [Freeswitch-users] How to get bridge call duration with lua In-Reply-To: References: Message-ID: Hi Michael, I want to customer call duration for billing details. ex: i am calling xxxxxxxxxx customer number , if customer call is hungup i want to customer call duration On Fri, Nov 14, 2014 at 11:45 PM, Michael Collins wrote: > Can you describe in a little more detail what you are trying to do? For > example, are you trying to get the bridge duration at some random point > during the call ("how long has this call been bridged so far?") or after > the call has ended? > > -MC > > On Fri, Nov 14, 2014 at 3:23 AM, jobin dcruz wrote: > >> Hi, >> >> I am using freeswitch bridge,every thing working with fine. >> >> >> > data="hangup_after_bridge=true"/> >> > data="ringback=${us-ring}"/> >> > data="sofia/gateway/outside/${customer_number}"/> >> >> My problem is >> >> How to get bridge call duration with lua? >> Cannot update CDR table with bridge call details.How to update it? >> >> -- >> Jobin D'cruz (LAMP Developer) >> Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com >> >> >> >> Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework >> Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI >> MooTools Smarty Firefox Extension >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jobin D'cruz (LAMP Developer) Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141115/b9d050b0/attachment.html From steveayre at gmail.com Sat Nov 15 13:58:47 2014 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 15 Nov 2014 10:58:47 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: I believe 12.04 is too old to be supported by FreeSWITCH, it might be worth upgrading to 14.04 Trusty (which is also a LTS release) and which I know FS builds fine on on. On 14 November 2014 21:47, Bruce Lefko wrote: > So I ran those two exports you gave me and added "-c precise" to my utils > call, but it still seems to be running into that pbuilder bug. Here's the > output or log/precise-amd64: > > pbuilder/build//cow.20008 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file > `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.20008 > -> Copying COW directory > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow > /var/cache/pbuilder/build//cow.22359 > I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist > failed, it didn't exist? > -> Invoking pbuilder > forking: pbuilder update --override-config --buildplace > /var/cache/pbuilder/build//cow.22359 --mirror > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise > --no-targz --internal-chrootexec chroot > /var/cache/pbuilder/build//cow.22359 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file > `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, > does not exist > > Thanks! > > > > On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko wrote: > >> Yeah, so it looks like in log/sid-amd64 I see the following over and over >> again: >> >> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >> -> Invoking pbuilder >> forking: pbuilder create --buildplace >> /var/cache/pbuilder/base-sid-amd64.cow --mirror >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >> --distribution sid --no-targz --extrapackages cowdancer >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Distribution is sid. >> I: Building the build environment >> I: running debootstrap >> /usr/sbin/debootstrap >> I: Retrieving Release >> E: Failed getting release file >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >> E: debootstrap failed >> W: Aborting with an error >> pbuilder create failed >> >> >> >> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >> wrote: >> >>> I am trying to tweak mod_spandsp in master and deb package the code for >>> use in my application. >>> >>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>> but it consistently gets stuck at "Building sid-amd64 debs.." >>> >>> First off, I'm not sure if this is the correct way to package custom >>> changes, but also why is the debian build stuck at this step? >>> >>> Thanks! >>> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141115/f95eee94/attachment.html From admin at blindi.net Sun Nov 16 15:24:08 2014 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 16 Nov 2014 13:24:08 +0100 (CET) Subject: [Freeswitch-users] Very bad voice fs and skype on quadcore In-Reply-To: References: Message-ID: Hi all, I use the actual solution on fs and skype The voice is very good to use a dual core processor. The voice is very bad, to use a Quadcore. I use a Debian 7 System. I set the parameters on the skype_extension: The voice is unchanged (very bad, for example a roboter) A voicetest: "this is my voice" "tss i voi". Is this a problem on pulseaudio? Can you help please? --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From robb at boardman.me.uk Mon Nov 17 04:05:52 2014 From: robb at boardman.me.uk (Robert Boardman) Date: Mon, 17 Nov 2014 01:05:52 +0000 Subject: [Freeswitch-users] Javascript scripting Message-ID: Hi, I cannot seem to get jsrun working on the windows win32 builds, the v8 modules dont seem to be there, is there something i'm missing? Thanks Robb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/0903a510/attachment.html From rajgopalfs at gmail.com Mon Nov 17 06:51:42 2014 From: rajgopalfs at gmail.com (raj) Date: Mon, 17 Nov 2014 03:51:42 +0000 (UTC) Subject: [Freeswitch-users] unable to do (raj) References: Message-ID: Kalyani Kulkarni writes: > > > > originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 > async full) > > > try 192.168.1.94:9050 async full, the colon is missing in your syntax? > > > Thanks for the response. Yes, you are correct but it was a typo when I posted here. Note that the actual command works and opens up a socket on 9050. The commands api and bgapi are not available. My observation is that "async" and "full" parameters are not being passed to mod esl in this scenario. I had to set them forcibly in the code to make them work. I may have to debug why its happening or simply add another "execute" command to set them. ( "execute" is working) This seems to be simple as I know where the code is. Also my observation is that I am facing the issue only if I am issuing the command in the inbound socket - and trying to control using api/bgapi from the resulting outbound socket. If the outbound socket is opened due to an incoming call with dial plan &socket... every this fine. I believe the parameters are being passed correctly. Any thoughts? Regars ________________________________________________________________________ _ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users > http://www.freeswitch.org From joelewhite at gmail.com Mon Nov 17 11:35:32 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 17 Nov 2014 03:35:32 -0500 Subject: [Freeswitch-users] Moving Voicemails Message-ID: I initially configured my FreeSWITCH with an IP address, but now for scalability and high availability I am changing to a domain name. I successfully changed the system over, but was wondering how do I move the voicemails to the new domain folder under voicemail? The obvious answer is to move the files, I already did that and restarted the system. The existing voicemails and greetings are not visible to FreeSWITCH when I call voicemail. How do I transfer them over? Thank you in advance for any light you may shed on this predicament -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/8db47aca/attachment.html From krice at freeswitch.org Mon Nov 17 12:04:14 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 17 Nov 2014 03:04:14 -0600 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: Message-ID: You need to update the database also On 11/17/14 2:35 AM, "Joel White" wrote: > I initially configured my FreeSWITCH with an IP address, but now for > scalability and high availability I am changing to a domain name.? I > successfully changed the system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did that and restarted the > system.? The existing voicemails and greetings are not visible to FreeSWITCH > when I call voicemail. > > How do I transfer them over?? > > Thank you in advance for any light you may shed on this predicament ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/4e15fa7f/attachment.html From k.mathy at hexanet.fr Mon Nov 17 12:12:36 2014 From: k.mathy at hexanet.fr (Kevin Mathy) Date: Mon, 17 Nov 2014 10:12:36 +0100 Subject: [Freeswitch-users] DTMF strange problems Message-ID: Hi, I'm facing a "strange" problem with DTMF, on a FS server I'm using (Version 1.5.8b git fe2a4d6 2014-02-14 00:03:11Z 64bit). For some calls going through FS, when customers send DTMF tones (using RFC2833), I see those logs : 2014-11-14 15:47:48.516849 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 1:742 > 2014-11-14 15:47:48.516849 [DEBUG] switch_channel.c:489 RECV DTMF 1:742 > 2014-11-14 15:47:49.056857 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF C:640 > 2014-11-14 15:47:49.056857 [DEBUG] switch_channel.c:489 RECV DTMF C:640 > 2014-11-14 15:47:49.356848 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 2:661 > 2014-11-14 15:47:49.356848 [DEBUG] switch_channel.c:489 RECV DTMF 2:661 > 2014-11-14 15:47:49.696846 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 3:665 > 2014-11-14 15:47:49.696846 [DEBUG] switch_channel.c:489 RECV DTMF 3:665 > 2014-11-14 15:47:50.256872 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 4:745 > 2014-11-14 15:47:50.256872 [DEBUG] switch_channel.c:489 RECV DTMF 4:745 > 2014-11-14 15:47:50.796855 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 5:741 > 2014-11-14 15:47:50.796855 [DEBUG] switch_channel.c:489 RECV DTMF 5:741 > 2014-11-14 15:47:51.236850 [DEBUG] switch_rtp.c:5683 RTP RECV DTMF 6:664 > 2014-11-14 15:47:51.236850 [DEBUG] switch_channel.c:489 RECV DTMF 6:664 > Here, the customer tried to send "123456". So, I don't know why this "C" character has been sent, and I'm sure the customer didn't send this, because there's no "C" touch on his phone's keypad... And this "C" results in a double "1" received on the IVR server (hosted on another server than FS). (Ex : 123456 => 1C23456 => 1123456) It doesn't happen everytime, but for around 80% of calls... So it's really a problem for us... I looked at PCAPs for concerned calls, and saw that DTMF tones are only sent with RFC2833, not with SIP INFO nor InBand. I first thought that digits may be received simultaneously with RFC2833 and InBand but definitely not. Do you think what could be the cause of that ? I don't use "start_dtmf" in my diaplans, I already saw in other ML topics that this may cause double digit problems. Is there a way to ask freeswitch to only handle 0-9*# characters ? Or any idea, I'll be glad to try anything that can help us ! Thanks for help, Kevin *Bien cordialement, Best Regards, **Kevin MATHY* | Ing?nieur VoIP -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/0da36f2f/attachment.html From steveayre at gmail.com Mon Nov 17 12:57:39 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 17 Nov 2014 09:57:39 +0000 Subject: [Freeswitch-users] DTMF strange problems In-Reply-To: References: Message-ID: > > I looked at PCAPs for concerned calls, and saw that DTMF tones are only > sent with RFC2833, not with SIP INFO nor InBand. > I first thought that digits may be received simultaneously with RFC2833 > and InBand but definitely not. Is that end-to-end, or are they reaching you via someone else? Or something else (ATA adapter)? If so perhaps some inband->RFC2833 is happening between you and them that's introducing the bad digit. If not can you share the PCAP (or just its RFC2833 packets)? That might explain what's happening. Do you think what could be the cause of that ? I don't use "start_dtmf" in > my diaplans, I already saw in other ML topics that this may cause double > digit problems. That's only if there's inband and out-of-band (RFC2833/INFO) DTMF at the same time. There's no issue otherwise, but using it on a call without inband DTMF is a waste of CPU. If you're getting RFC2833 then any inband audio will only cause double digits if you run start_dtmf, inband will be ignored by default. Is there a way to ask freeswitch to only handle 0-9*# characters ? Or any > idea, I'll be glad to try anything that can help us ! The problem is RFC2833 will be saying the 1 digit finished before the C one started, FS can only interpret it as its told to. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/7fbc8ed3/attachment-0001.html From lists at kavun.ch Mon Nov 17 15:07:00 2014 From: lists at kavun.ch (Emrah) Date: Mon, 17 Nov 2014 13:07:00 +0100 Subject: [Freeswitch-users] TLS setup failed In-Reply-To: References: Message-ID: <94F8D135-0133-48C5-8D4C-B5CC46219CEB@kavun.ch> Hi all, I tried regenerating new certificates with a lower key size, bypassed the commercial component and instead uploaded the CA on my devices, changed the SSL port and? Well still stuck. Your help is more than welcome! Any idea? Emrah > On Nov 14, 2014, at 5:14 PM, Emrah wrote: > > Hi list, > > I am able to use FS with SSLv23 with Blink Pro on Mac OS, but that?s about it. > I get the following error if I connect with any other device (Bria iOS, Yealink phone, Join Softphone): > TLS setup failed (error:00000001:lib(0):func(0):reason(1)) > > I came across this thread: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-td7587736.html but it doesn?t seem to apply to my scenario. > > I am using a commercial certificate. My devices connect to a domain which has an SRV record which points to itself on the SSL port. SSL host is an A record and matches the CN in the certificate. Server cA check is even turned off on certain phones. > > The only error I get on my Yealink phone is this: > Nov 14 11:04:03 SIP [524]: SDL <6+info > [000] SSL_connect (read done) > Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL ERROR > Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL_connect error > > I would appreciate to know how I could debug this further. Or if you have any clue at what may be going on. > > Thanks! > Emrah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/82a8bcbb/attachment.html From avi at avimarcus.net Mon Nov 17 15:22:29 2014 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 17 Nov 2014 12:22:29 +0000 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: References: Message-ID: <00000149bdb44840-13848616-cbcd-4b5f-8431-59cf8fc691ac-000000@email.amazonses.com> The path isn't dynamically generated based on the domain, it's stored per-file in the database. You can update the db or symlink the folders so they are available in the old structure too. -Avi On Mon, Nov 17, 2014 at 10:35 AM, Joel White wrote: > I initially configured my FreeSWITCH with an IP address, but now for > scalability and high availability I am changing to a domain name. I > successfully changed the system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did that and restarted > the system. The existing voicemails and greetings are not visible to > FreeSWITCH when I call voicemail. > > How do I transfer them over? > > Thank you in advance for any light you may shed on this predicament > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/3d84bce1/attachment.html From akhilgarg7 at gmail.com Mon Nov 17 15:38:12 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Mon, 17 Nov 2014 18:08:12 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: Hi, I am using 1.5.4b freeswitch release. I am running freeswitch by commands "freeswtich" or "freeswtich -nc" running fs_cli hangs. it gives no output. I could see the socket is open by running command "netstat -anlp | grep 8021" "tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 3084/freeswitch" Please help -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/dea43bc5/attachment.html From rtreleaven at bunnykick.ca Mon Nov 17 15:47:29 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Mon, 17 Nov 2014 07:47:29 -0500 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: try "./freeswitch -nonat" On Mon, Nov 17, 2014 at 7:38 AM, akhil garg wrote: > Hi, > I am using 1.5.4b freeswitch release. > > I am running freeswitch by commands "freeswtich" or "freeswtich -nc" > > running fs_cli hangs. > it gives no output. > > I could see the socket is open by running command "netstat -anlp | grep > 8021" > "tcp 0 0 127.0.0.1:8021 0.0.0.0:* > LISTEN 3084/freeswitch" > > > > Please help > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/3cb1de63/attachment.html From akhilgarg7 at gmail.com Mon Nov 17 15:49:05 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Mon, 17 Nov 2014 18:19:05 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: sorry, I am using freeswitch version 1.5.6b On Mon, Nov 17, 2014 at 6:08 PM, akhil garg wrote: > Hi, > I am using 1.5.4b freeswitch release. > > I am running freeswitch by commands "freeswtich" or "freeswtich -nc" > > running fs_cli hangs. > it gives no output. > > I could see the socket is open by running command "netstat -anlp | grep > 8021" > "tcp 0 0 127.0.0.1:8021 0.0.0.0:* > LISTEN 3084/freeswitch" > > > > Please help > > -- > regards, > akhil > -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/7cf282dd/attachment.html From akhilgarg7 at gmail.com Mon Nov 17 16:01:50 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Mon, 17 Nov 2014 18:31:50 +0530 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: fs_cli still not opening after running "./freeswitch -nonat" Regards, Akhil On Mon, Nov 17, 2014 at 6:19 PM, akhil garg wrote: > sorry, I am using freeswitch version 1.5.6b > > On Mon, Nov 17, 2014 at 6:08 PM, akhil garg wrote: > >> Hi, >> I am using 1.5.4b freeswitch release. >> >> I am running freeswitch by commands "freeswtich" or "freeswtich -nc" >> >> running fs_cli hangs. >> it gives no output. >> >> I could see the socket is open by running command "netstat -anlp | grep >> 8021" >> "tcp 0 0 127.0.0.1:8021 0.0.0.0:* >> LISTEN 3084/freeswitch" >> >> >> >> Please help >> >> -- >> regards, >> akhil >> > > > > -- > regards, > akhil > -- regards, akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/e7cb5874/attachment-0001.html From mvar78 at gmail.com Mon Nov 17 16:34:06 2014 From: mvar78 at gmail.com (Massimo Varriale) Date: Mon, 17 Nov 2014 14:34:06 +0100 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Then I suggest you to simply run the command "freeswitch" without any parameters so you can see at the startup console why the Freeswitch process is hanging up. It could be a wrong parameter into some XML file or whatever.. BTW, why not using a stable version (I mean 1.4) rather than a development version? Cheers Max Il giorno 17/nov/2014, alle ore 14:01, akhil garg ha scritto: > fs_cli still not opening after running "./freeswitch -nonat" > > > > Regards, > Akhil > > On Mon, Nov 17, 2014 at 6:19 PM, akhil garg wrote: > sorry, I am using freeswitch version 1.5.6b > > On Mon, Nov 17, 2014 at 6:08 PM, akhil garg wrote: > Hi, > I am using 1.5.4b freeswitch release. > > I am running freeswitch by commands "freeswtich" or "freeswtich -nc" > > running fs_cli hangs. > it gives no output. > > I could see the socket is open by running command "netstat -anlp | grep 8021" > "tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 3084/freeswitch" > > > > Please help > > -- > regards, > akhil > > > > -- > regards, > akhil > > > > -- > regards, > akhil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/c8520e6b/attachment.html From brian at freeswitch.org Mon Nov 17 16:35:42 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:35:42 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: How about trying some code thats a bit more recent than 1.5.6b? We're up to 1.5.15b on master, and 1.4.13 on release. On Mon, Nov 17, 2014 at 7:01 AM, akhil garg wrote: > fs_cli still not opening after running "./freeswitch -nonat" > > > > Regards, > Akhil > > On Mon, Nov 17, 2014 at 6:19 PM, akhil garg wrote: > >> sorry, I am using freeswitch version 1.5.6b >> >> On Mon, Nov 17, 2014 at 6:08 PM, akhil garg wrote: >> >>> Hi, >>> I am using 1.5.4b freeswitch release. >>> >>> I am running freeswitch by commands "freeswtich" or "freeswtich -nc" >>> >>> running fs_cli hangs. >>> it gives no output. >>> >>> I could see the socket is open by running command "netstat -anlp | grep >>> 8021" >>> "tcp 0 0 127.0.0.1:8021 0.0.0.0:* >>> LISTEN 3084/freeswitch" >>> >>> >>> >>> Please help >>> >>> -- >>> regards, >>> akhil >>> >> >> >> >> -- >> regards, >> akhil >> > > > > -- > regards, > akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/47c8cb26/attachment.html From brian at freeswitch.org Mon Nov 17 16:42:06 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:42:06 -0600 Subject: [Freeswitch-users] TLS setup failed In-Reply-To: <94F8D135-0133-48C5-8D4C-B5CC46219CEB@kavun.ch> References: <94F8D135-0133-48C5-8D4C-B5CC46219CEB@kavun.ch> Message-ID: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/docs/how_to_make_your_own_ca_correctly.txt?at=79b3cdfc967376511d113d6386e450a5f7ab0db2&raw On Mon, Nov 17, 2014 at 6:07 AM, Emrah wrote: > Hi all, > > I tried regenerating new certificates with a lower key size, bypassed the > commercial component and instead uploaded the CA on my devices, changed the > SSL port and? Well still stuck. > > Your help is more than welcome! > > Any idea? > > Emrah > > > On Nov 14, 2014, at 5:14 PM, Emrah wrote: > > Hi list, > > I am able to use FS with SSLv23 with Blink Pro on Mac OS, but that?s about > it. > I get the following error if I connect with any other device (Bria iOS, > Yealink phone, Join Softphone): > TLS setup failed (error:00000001:lib(0):func(0):reason(1)) > > I came across this thread: > http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-td7587736.html > but it doesn?t seem to apply to my scenario. > > I am using a commercial certificate. My devices connect to a domain which > has an SRV record which points to itself on the SSL port. SSL host is an A > record and matches the CN in the certificate. Server cA check is even > turned off on certain phones. > > The only error I get on my Yealink phone is this: > Nov 14 11:04:03 SIP [524]: SDL <6+info > [000] SSL_connect (read done) > Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL ERROR > Nov 14 11:04:03 SIP [524]: SDL <3+error > [000] SSL_connect error > > I would appreciate to know how I could debug this further. Or if you have > any clue at what may be going on. > > Thanks! > Emrah > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/c23770a6/attachment.html From brian at freeswitch.org Mon Nov 17 16:43:03 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:43:03 -0600 Subject: [Freeswitch-users] DTMF strange problems In-Reply-To: References: Message-ID: Dialing on speaker phone maybe? On Mon, Nov 17, 2014 at 3:57 AM, Steven Ayre wrote: > I looked at PCAPs for concerned calls, and saw that DTMF tones are only >> sent with RFC2833, not with SIP INFO nor InBand. >> I first thought that digits may be received simultaneously with RFC2833 >> and InBand but definitely not. > > > Is that end-to-end, or are they reaching you via someone else? Or > something else (ATA adapter)? If so perhaps some inband->RFC2833 is > happening between you and them that's introducing the bad digit. > > If not can you share the PCAP (or just its RFC2833 packets)? That might > explain what's happening. > > > Do you think what could be the cause of that ? I don't use "start_dtmf" >> in my diaplans, I already saw in other ML topics that this may cause double >> digit problems. > > > That's only if there's inband and out-of-band (RFC2833/INFO) DTMF at the > same time. There's no issue otherwise, but using it on a call without > inband DTMF is a waste of CPU. > > If you're getting RFC2833 then any inband audio will only cause double > digits if you run start_dtmf, inband will be ignored by default. > > > Is there a way to ask freeswitch to only handle 0-9*# characters ? Or any >> idea, I'll be glad to try anything that can help us ! > > > The problem is RFC2833 will be saying the 1 digit finished before the C > one started, FS can only interpret it as its told to. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/8b683362/attachment-0001.html From brian at freeswitch.org Mon Nov 17 16:44:04 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:44:04 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Or its that evil single core spin forever bug we fixed long ago, this is why you should always test on master prior to posting to the list to verify that its still happening. On Mon, Nov 17, 2014 at 7:34 AM, Massimo Varriale wrote: > Then I suggest you to simply run the command "freeswitch" without any > parameters so you can see at the startup console why the Freeswitch process > is hanging up. > > It could be a wrong parameter into some XML file or whatever.. > > > BTW, why not using a stable version (I mean 1.4) rather than a development > version? > > Cheers > Max > > > > > Il giorno 17/nov/2014, alle ore 14:01, akhil garg ha scritto: > > fs_cli still not opening after running "./freeswitch -nonat" > > > > Regards, > Akhil > > On Mon, Nov 17, 2014 at 6:19 PM, akhil garg wrote: > >> sorry, I am using freeswitch version 1.5.6b >> >> On Mon, Nov 17, 2014 at 6:08 PM, akhil garg wrote: >> >>> Hi, >>> I am using 1.5.4b freeswitch release. >>> >>> I am running freeswitch by commands "freeswtich" or "freeswtich -nc" >>> >>> running fs_cli hangs. >>> it gives no output. >>> >>> I could see the socket is open by running command "netstat -anlp | grep >>> 8021" >>> "tcp 0 0 127.0.0.1:8021 0.0.0.0:* >>> LISTEN 3084/freeswitch" >>> >>> >>> >>> Please help >>> >>> -- >>> regards, >>> akhil >>> >> >> >> >> -- >> regards, >> akhil >> > > > > -- > regards, > akhil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/b3f1003f/attachment.html From brian at freeswitch.org Mon Nov 17 16:45:43 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:45:43 -0600 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: Is the module loaded? 'module_exists mod_v8', see if that returns true. Else try 'load mod_v8', see if its just not loaded. On Sun, Nov 16, 2014 at 7:05 PM, Robert Boardman wrote: > Hi, I cannot seem to get jsrun working on the windows win32 builds, the v8 > modules dont seem to be there, is there something i'm missing? > > Thanks > > Robb > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/de0dc231/attachment.html From mike at jerris.com Mon Nov 17 16:46:47 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 08:46:47 -0500 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: Build on 12.04 generally works if you fix all the dependency issues (there are several). That being said, we have had a string of well known very bad behaviors due to bugs in 12.04. We strongly recommend against using it. You will have random crashes and we will not spend any time trying to fix them due to these known issues. > On Nov 15, 2014, at 5:58 AM, Steven Ayre wrote: > > I believe 12.04 is too old to be supported by FreeSWITCH, it might be worth upgrading to 14.04 Trusty (which is also a LTS release) and which I know FS builds fine on on. > > On 14 November 2014 21:47, Bruce Lefko > wrote: > So I ran those two exports you gave me and added "-c precise" to my utils call, but it still seems to be running into that pbuilder bug. Here's the output or log/precise-amd64: > > pbuilder/build//cow.20008 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.20008 > -> Copying COW directory > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow /var/cache/pbuilder/build//cow.22359 > I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist failed, it didn't exist? > -> Invoking pbuilder > forking: pbuilder update --override-config --buildplace /var/cache/pbuilder/build//cow.22359 --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise --no-targz --internal-chrootexec chroot /var/cache/pbuilder/build//cow.22359 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, does not exist > > Thanks! > > > > On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko > wrote: > Yeah, so it looks like in log/sid-amd64 I see the following over and over again: > > forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow > -> Invoking pbuilder > forking: pbuilder create --buildplace /var/cache/pbuilder/base-sid-amd64.cow --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 --distribution sid --no-targz --extrapackages cowdancer > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Distribution is sid. > I: Building the build environment > I: running debootstrap > /usr/sbin/debootstrap > I: Retrieving Release > E: Failed getting release file http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release > E: debootstrap failed > W: Aborting with an error > pbuilder create failed > > > > On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko > wrote: > I am trying to tweak mod_spandsp in master and deb package the code for use in my application. > > I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but it consistently gets stuck at "Building sid-amd64 debs.." > > First off, I'm not sure if this is the correct way to package custom changes, but also why is the debian build stuck at this step? > > Thanks! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/71a028e1/attachment-0001.html From brian at freeswitch.org Mon Nov 17 16:47:57 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 07:47:57 -0600 Subject: [Freeswitch-users] How to get bridge call duration with lua In-Reply-To: References: Message-ID: tsk tsk tsk, you really shouldn't be doing billing inline with your call flow, There are many ways of accomplishing this without having to invade the session itself to accomplish billing. Is that all you're trying to do is billing calls? Can you elaborate a little bit more? On Fri, Nov 14, 2014 at 8:48 PM, jobin dcruz wrote: > Hi Michael, > > I want to customer call duration for billing details. > > > > ex: i am calling xxxxxxxxxx customer number , if customer call is hungup i > want to customer call duration > > > On Fri, Nov 14, 2014 at 11:45 PM, Michael Collins > wrote: > >> Can you describe in a little more detail what you are trying to do? For >> example, are you trying to get the bridge duration at some random point >> during the call ("how long has this call been bridged so far?") or after >> the call has ended? >> >> -MC >> >> On Fri, Nov 14, 2014 at 3:23 AM, jobin dcruz >> wrote: >> >>> Hi, >>> >>> I am using freeswitch bridge,every thing working with fine. >>> >>> >>> >> data="hangup_after_bridge=true"/> >>> >> data="ringback=${us-ring}"/> >>> >> data="sofia/gateway/outside/${customer_number}"/> >>> >>> My problem is >>> >>> How to get bridge call duration with lua? >>> Cannot update CDR table with bridge call details.How to update it? >>> >>> -- >>> Jobin D'cruz (LAMP Developer) >>> Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com >>> >>> >>> >>> Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework >>> Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI >>> MooTools Smarty Firefox Extension >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Jobin D'cruz (LAMP Developer) > Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com > > > > Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework > Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI > MooTools Smarty Firefox Extension > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/fcb43a33/attachment.html From mike at jerris.com Mon Nov 17 16:49:04 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 08:49:04 -0500 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: <2D242A85-6255-467A-8FE9-0DEBF6C41B95@jerris.com> We may have never bothered to get these to build on 32 bit. Is there a reason you can't use the 64 bit build? If not, i would strongly recommend using that instead. > On Nov 16, 2014, at 8:05 PM, Robert Boardman wrote: > > Hi, I cannot seem to get jsrun working on the windows win32 builds, the v8 modules dont seem to be there, is there something i'm missing? > From alex at digitalmail.com Mon Nov 17 16:53:54 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 17 Nov 2014 13:53:54 +0000 Subject: [Freeswitch-users] Freeswitch db for voicemail Message-ID: <5469FDF2.8060803@digitalmail.com> Where does one configure the database that the voicemail system uses? Doesn't seem to be using the odbc/mysql database that I specified for core, but I'm not sure if voicemail is supposed to be part of core? From steveayre at gmail.com Mon Nov 17 16:58:59 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 17 Nov 2014 13:58:59 +0000 Subject: [Freeswitch-users] Freeswitch db for voicemail In-Reply-To: <5469FDF2.8060803@digitalmail.com> References: <5469FDF2.8060803@digitalmail.com> Message-ID: odbc-dsn in mod_voicemail.conf.xml Generally modules that use database have their own DSN, which may or may not be the same as the core. You want to configure them all separately. For example mod_sofia has a odbc-dsn for each profile. Everything will use sqlite by default (because there are no dependancies), regardless of what the core is using. On 17 November 2014 13:53, Alex Lake wrote: > Where does one configure the database that the voicemail system uses? > Doesn't seem to be using the odbc/mysql database that I specified for > core, but I'm not sure if voicemail is supposed to be part of core? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/0988cff0/attachment.html From mike at jerris.com Mon Nov 17 17:00:07 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 09:00:07 -0500 Subject: [Freeswitch-users] Freeswitch db for voicemail In-Reply-To: <5469FDF2.8060803@digitalmail.com> References: <5469FDF2.8060803@digitalmail.com> Message-ID: <9E5F924B-ED95-4D33-AE1C-213377E5525C@jerris.com> https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/conf/vanilla/autoload_configs/voicemail.conf.xml#68 > On Nov 17, 2014, at 8:53 AM, Alex Lake wrote: > > Where does one configure the database that the voicemail system uses? > Doesn't seem to be using the odbc/mysql database that I specified for > core, but I'm not sure if voicemail is supposed to be part of core? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/e9a2cade/attachment-0001.html From steveayre at gmail.com Mon Nov 17 17:01:44 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 17 Nov 2014 14:01:44 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: > > That being said, we have had a string of well known very bad behaviors due > to bugs in 12.04 Can you elaborate Michael? (The company I work for runs FS exclusively on servers running 12.04, although my personal preference would be Debian). On 17 November 2014 13:46, Michael Jerris wrote: > Build on 12.04 generally works if you fix all the dependency issues (there > are several). That being said, we have had a string of well known very bad > behaviors due to bugs in 12.04. We strongly recommend against using it. > You will have random crashes and we will not spend any time trying to fix > them due to these known issues. > > > On Nov 15, 2014, at 5:58 AM, Steven Ayre wrote: > > I believe 12.04 is too old to be supported by FreeSWITCH, it might be > worth upgrading to 14.04 Trusty (which is also a LTS release) and which I > know FS builds fine on on. > > On 14 November 2014 21:47, Bruce Lefko wrote: > >> So I ran those two exports you gave me and added "-c precise" to my utils >> call, but it still seems to be running into that pbuilder bug. Here's the >> output or log/precise-amd64: >> >> pbuilder/build//cow.20008 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file >> `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.20008 >> -> Copying COW directory >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow >> /var/cache/pbuilder/build//cow.22359 >> I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist >> failed, it didn't exist? >> -> Invoking pbuilder >> forking: pbuilder update --override-config --buildplace >> /var/cache/pbuilder/build//cow.22359 --mirror >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise >> --no-targz --internal-chrootexec chroot >> /var/cache/pbuilder/build//cow.22359 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file >> `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, >> does not exist >> >> Thanks! >> >> >> >> On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko >> wrote: >> >>> Yeah, so it looks like in log/sid-amd64 I see the following over and >>> over again: >>> >>> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >>> -> Invoking pbuilder >>> forking: pbuilder create --buildplace >>> /var/cache/pbuilder/base-sid-amd64.cow --mirror >>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >>> --distribution sid --no-targz --extrapackages cowdancer >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Distribution is sid. >>> I: Building the build environment >>> I: running debootstrap >>> /usr/sbin/debootstrap >>> I: Retrieving Release >>> E: Failed getting release file >>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >>> E: debootstrap failed >>> W: Aborting with an error >>> pbuilder create failed >>> >>> >>> >>> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >>> wrote: >>> >>>> I am trying to tweak mod_spandsp in master and deb package the code for >>>> use in my application. >>>> >>>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>>> but it consistently gets stuck at "Building sid-amd64 debs.." >>>> >>>> First off, I'm not sure if this is the correct way to package custom >>>> changes, but also why is the debian build stuck at this step? >>>> >>>> Thanks! >>>> >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/f4f820b6/attachment.html From alex at digitalmail.com Mon Nov 17 17:03:52 2014 From: alex at digitalmail.com (Alex Lake) Date: Mon, 17 Nov 2014 14:03:52 +0000 Subject: [Freeswitch-users] Freeswitch db for voicemail In-Reply-To: <5469FDF2.8060803@digitalmail.com> References: <5469FDF2.8060803@digitalmail.com> Message-ID: <546A0048.6080900@digitalmail.com> Thanks all for the speedy responses. A little embarassing as I found the answer myself just after posting the question! Often seems to be the way that the answer is in a correctly worded question... 42! Alex > Where does one configure the database that the voicemail system uses? > Doesn't seem to be using the odbc/mysql database that I specified for > core, but I'm not sure if voicemail is supposed to be part of core? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2014.0.4765 / Virus Database: 4189/8565 - Release Date: 11/13/14 > > From sdame at 207me.com Mon Nov 17 17:16:26 2014 From: sdame at 207me.com (Stephen Dame) Date: Mon, 17 Nov 2014 09:16:26 -0500 Subject: [Freeswitch-users] mod_conference stereo? record stereo wav? Message-ID: <04e901d00271$13bae5c0$3b30b140$@207me.com> Question, we are successfully connecting chrome/firefox with opus48, 2 channel stereo to mod_conference user master build. Recording the conference seems to only record 1 channel Couple questions, did not see any documentation on stereo in wiki. 1) Do we need to add a param to conference.conf.xml profile to have the conference mix stereo? 2) What is the proper way to record the conference in stereo wav file? Thanks in advance. Regards, Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/48453e16/attachment.html From mike at jerris.com Mon Nov 17 17:25:05 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 09:25:05 -0500 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: <5F2EF027-BFA2-4ACF-A08A-12D50083DED6@jerris.com> Random crashes. We have traced a number of them down to changes/patches pulled in to system libs, but stopped bothering, its not worth the time as the new LTS works fine. > On Nov 17, 2014, at 9:01 AM, Steven Ayre wrote: > > That being said, we have had a string of well known very bad behaviors due to bugs in 12.04 > > Can you elaborate Michael? (The company I work for runs FS exclusively on servers running 12.04, although my personal preference would be Debian). > > On 17 November 2014 13:46, Michael Jerris > wrote: > Build on 12.04 generally works if you fix all the dependency issues (there are several). That being said, we have had a string of well known very bad behaviors due to bugs in 12.04. We strongly recommend against using it. You will have random crashes and we will not spend any time trying to fix them due to these known issues. > > >> On Nov 15, 2014, at 5:58 AM, Steven Ayre > wrote: >> >> I believe 12.04 is too old to be supported by FreeSWITCH, it might be worth upgrading to 14.04 Trusty (which is also a LTS release) and which I know FS builds fine on on. >> >> On 14 November 2014 21:47, Bruce Lefko > wrote: >> So I ran those two exports you gave me and added "-c precise" to my utils call, but it still seems to be running into that pbuilder bug. Here's the output or log/precise-amd64: >> >> pbuilder/build//cow.20008 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.20008 >> -> Copying COW directory >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow /var/cache/pbuilder/build//cow.22359 >> I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist failed, it didn't exist? >> -> Invoking pbuilder >> forking: pbuilder update --override-config --buildplace /var/cache/pbuilder/build//cow.22359 --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise --no-targz --internal-chrootexec chroot /var/cache/pbuilder/build//cow.22359 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, does not exist >> >> Thanks! >> >> >> >> On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko > wrote: >> Yeah, so it looks like in log/sid-amd64 I see the following over and over again: >> >> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >> -> Invoking pbuilder >> forking: pbuilder create --buildplace /var/cache/pbuilder/base-sid-amd64.cow --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 --distribution sid --no-targz --extrapackages cowdancer >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Distribution is sid. >> I: Building the build environment >> I: running debootstrap >> /usr/sbin/debootstrap >> I: Retrieving Release >> E: Failed getting release file http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >> E: debootstrap failed >> W: Aborting with an error >> pbuilder create failed >> >> >> >> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko > wrote: >> I am trying to tweak mod_spandsp in master and deb package the code for use in my application. >> >> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but it consistently gets stuck at "Building sid-amd64 debs.." >> >> First off, I'm not sure if this is the correct way to package custom changes, but also why is the debian build stuck at this step? >> >> Thanks! >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/aee877c1/attachment-0001.html From jason.holden at start.ca Mon Nov 17 19:11:12 2014 From: jason.holden at start.ca (Jason Holden) Date: Mon, 17 Nov 2014 11:11:12 -0500 Subject: [Freeswitch-users] lua script to disconnect active outbound calls to a specific gateway Message-ID: <88A6AB16FA7E42F88178F237CAF0D97B@bob> Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/c7da71a1/attachment.html From joelewhite at gmail.com Mon Nov 17 19:07:20 2014 From: joelewhite at gmail.com (Joel White) Date: Mon, 17 Nov 2014 11:07:20 -0500 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: <00000149bdb44840-13848616-cbcd-4b5f-8431-59cf8fc691ac-000000@email.amazonses.com> References: <00000149bdb44840-13848616-cbcd-4b5f-8431-59cf8fc691ac-000000@email.amazonses.com> Message-ID: What is the easiest way to make this change? SQLite3 via command line? Or is there another way to manipulate the database? On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: > The path isn't dynamically generated based on the domain, it's stored > per-file in the database. > You can update the db or symlink the folders so they are available in the > old structure too. > -Avi > > On Mon, Nov 17, 2014 at 10:35 AM, Joel White wrote: > >> I initially configured my FreeSWITCH with an IP address, but now for >> scalability and high availability I am changing to a domain name. I >> successfully changed the system over, but was wondering how do I move the >> voicemails to the new domain folder under voicemail? >> >> The obvious answer is to move the files, I already did that and restarted >> the system. The existing voicemails and greetings are not visible to >> FreeSWITCH when I call voicemail. >> >> How do I transfer them over? >> >> Thank you in advance for any light you may shed on this predicament >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/cd5efa7e/attachment.html From jobindcruz at gmail.com Mon Nov 17 20:54:49 2014 From: jobindcruz at gmail.com (jobin dcruz) Date: Mon, 17 Nov 2014 23:24:49 +0530 Subject: [Freeswitch-users] How to get bridge call duration with lua In-Reply-To: References: Message-ID: Issue is solved.Thanks all On Mon, Nov 17, 2014 at 7:17 PM, Brian West wrote: > tsk tsk tsk, you really shouldn't be doing billing inline with your call > flow, There are many ways of accomplishing this without having to invade > the session itself to accomplish billing. Is that all you're trying to do > is billing calls? Can you elaborate a little bit more? > > On Fri, Nov 14, 2014 at 8:48 PM, jobin dcruz wrote: > >> Hi Michael, >> >> I want to customer call duration for billing details. >> >> >> >> ex: i am calling xxxxxxxxxx customer number , if customer call is hungup >> i want to customer call duration >> >> >> On Fri, Nov 14, 2014 at 11:45 PM, Michael Collins >> wrote: >> >>> Can you describe in a little more detail what you are trying to do? For >>> example, are you trying to get the bridge duration at some random point >>> during the call ("how long has this call been bridged so far?") or after >>> the call has ended? >>> >>> -MC >>> >>> On Fri, Nov 14, 2014 at 3:23 AM, jobin dcruz >>> wrote: >>> >>>> Hi, >>>> >>>> I am using freeswitch bridge,every thing working with fine. >>>> >>>> >>>> >>> data="hangup_after_bridge=true"/> >>>> >>> data="ringback=${us-ring}"/> >>>> >>> data="sofia/gateway/outside/${customer_number}"/> >>>> >>>> My problem is >>>> >>>> How to get bridge call duration with lua? >>>> Cannot update CDR table with bridge call details.How to update it? >>>> >>>> -- >>>> Jobin D'cruz (LAMP Developer) >>>> Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com >>>> >>>> >>>> >>>> Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework >>>> Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI >>>> MooTools Smarty Firefox Extension >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Jobin D'cruz (LAMP Developer) >> Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com >> >> >> >> Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework >> Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI >> MooTools Smarty Firefox Extension >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Jobin D'cruz (LAMP Developer) Email: jobindcruz at gmail.com | lampdeveloper.jobindcruz at gmail.com Php MySql Linux Apache Ajax FFmpeg FFmpeg-php Mplayer Zend Framework Codeigniter CakePhp Wordpress Drupal Joomla Html5 Css3 jQuery YUI MooTools Smarty Firefox Extension -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/c84e4c1a/attachment-0001.html From krice at freeswitch.org Mon Nov 17 21:08:12 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 17 Nov 2014 12:08:12 -0600 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: Message-ID: You got it sqlite3 is the way to go... I doubt doing what you are trying is heavily documented tho as its not something normally done. But all metadata for voicemail is in the sqlite db and that includes file paths etc On 11/17/14 10:07 AM, "Joel White" wrote: > What is the easiest way to make this change? > > SQLite3 via command line?? Or is there another way to manipulate the database? > > On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: >> The path isn't dynamically generated based on the domain, it's stored >> per-file in the database. >> You can update the db or symlink the folders so they are available in the old >> structure too. >> -Avi >> >> On Mon, Nov 17, 2014 at 10:35 AM, Joel White wrote: >>> I initially configured my FreeSWITCH with an IP address, but now for >>> scalability and high availability I am changing to a domain name.? I >>> successfully changed the system over, but was wondering how do I move the >>> voicemails to the new domain folder under voicemail? >>> >>> The obvious answer is to move the files, I already did that and restarted >>> the system.? The existing voicemails and greetings are not visible to >>> FreeSWITCH when I call voicemail. >>> >>> How do I transfer them over?? >>> >>> Thank you in advance for any light you may shed on this predicament ? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/fc98c494/attachment.html From pnlsantos at gmail.com Mon Nov 17 21:08:58 2014 From: pnlsantos at gmail.com (Pedro Santos) Date: Mon, 17 Nov 2014 18:08:58 +0000 Subject: [Freeswitch-users] perl esl Message-ID: Hi, i can not install the perlmod esl. There is no perl/ESL.so file. i am using centos 6 and the stable version from git of freeswitch. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/e409eed1/attachment.html From pnlsantos at gmail.com Mon Nov 17 21:19:08 2014 From: pnlsantos at gmail.com (Pedro Santos) Date: Mon, 17 Nov 2014 18:19:08 +0000 Subject: [Freeswitch-users] perl esl In-Reply-To: References: Message-ID: FreeSWITCH Version 1.5.15b+git~20141112T053130Z~07030c63f0~64bit (git 07030c6 2014-11-12 05:31:30Z 64bit) On Mon, Nov 17, 2014 at 6:08 PM, Pedro Santos wrote: > Hi, > > i can not install the perlmod esl. > There is no perl/ESL.so file. > i am using centos 6 and the stable version from git of freeswitch. > > Regards > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/5ea4669c/attachment.html From darren at aleph-com.net Mon Nov 17 21:39:43 2014 From: darren at aleph-com.net (Darren Wiebe) Date: Mon, 17 Nov 2014 11:39:43 -0700 Subject: [Freeswitch-users] perl esl In-Reply-To: References: Message-ID: Brian helped on IRC on Saturday with the same question for Debian. On Centos try "yum install perl-devel" Darren Wiebe On Mon, Nov 17, 2014 at 11:08 AM, Pedro Santos wrote: > Hi, > > i can not install the perlmod esl. > There is no perl/ESL.so file. > i am using centos 6 and the stable version from git of freeswitch. > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/bdaaa1ae/attachment.html From freeswitch at earthspike.net Mon Nov 17 21:44:37 2014 From: freeswitch at earthspike.net (John) Date: Mon, 17 Nov 2014 18:44:37 +0000 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: References: Message-ID: <546A4215.2090509@earthspike.net> I've done this and for exactly the same reason. I moved the files to the domain-named path and updated the database using sqlite3. I found this in my command history, but you should test first on a copy of the db. It's for upgrading from 1.2.quite_old to 1.4.quite_recent. $ sqlite3 voicemail_default.db sqlite> update voicemail_msgs set file_path='/var/lib' || substr(replace(file_path,'172.16.12.34','pbx.somewhere.net'),11); sqlite> update voicemail_msgs set file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net'); sqlite> update voicemail_prefs set greeting_path='/var/lib' || substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net'),11); I cannot remember what the '11' was for; I've changed the IP address and FQDN so you might need to tune this value; check with SELECTs in place of the UPDATEs. John On 17/11/14 18:08, Ken Rice wrote: > Re: [Freeswitch-users] Moving Voicemails You got it sqlite3 is the way > to go... I doubt doing what you are trying is heavily documented tho > as its not something normally done. > > But all metadata for voicemail is in the sqlite db and that includes > file paths etc > > > On 11/17/14 10:07 AM, "Joel White" wrote: > > What is the easiest way to make this change? > > SQLite3 via command line? Or is there another way to manipulate > the database? > > On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: > > The path isn't dynamically generated based on the domain, it's > stored per-file in the database. > You can update the db or symlink the folders so they are > available in the old structure too. > -Avi > > On Mon, Nov 17, 2014 at 10:35 AM, Joel White > wrote: > > I initially configured my FreeSWITCH with an IP address, > but now for scalability and high availability I am > changing to a domain name. I successfully changed the > system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did > that and restarted the system. The existing voicemails > and greetings are not visible to FreeSWITCH when I call > voicemail. > > How do I transfer them over? > > Thank you in advance for any light you may shed on this > predicament > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/e5d9b4ff/attachment-0001.html From mike at jerris.com Mon Nov 17 21:56:35 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 13:56:35 -0500 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: <546A4215.2090509@earthspike.net> References: <546A4215.2090509@earthspike.net> Message-ID: <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> This does seem kind of clumsy... I'd be happy to accept a patch for mod_voicemail change domain that moves files and updates db correctly if anyone wants to give it a try. > On Nov 17, 2014, at 1:44 PM, John wrote: > > I've done this and for exactly the same reason. I moved the files to the domain-named path and updated the database using sqlite3. I found this in my command history, but you should test first on a copy of the db. It's for upgrading from 1.2.quite_old to 1.4.quite_recent. > > $ sqlite3 voicemail_default.db > sqlite> update voicemail_msgs set file_path='/var/lib' || substr(replace(file_path,'172.16.12.34','pbx.somewhere.net '),11); > sqlite> update voicemail_msgs set file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net '); > sqlite> update voicemail_prefs set greeting_path='/var/lib' || substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net '),11); > > I cannot remember what the '11' was for; I've changed the IP address and FQDN so you might need to tune this value; check with SELECTs in place of the UPDATEs. > > John > > On 17/11/14 18:08, Ken Rice wrote: >> You got it sqlite3 is the way to go... I doubt doing what you are trying is heavily documented tho as its not something normally done. >> >> But all metadata for voicemail is in the sqlite db and that includes file paths etc >> >> >> On 11/17/14 10:07 AM, "Joel White" > wrote: >> >> What is the easiest way to make this change? >> >> SQLite3 via command line? Or is there another way to manipulate the database? >> >> On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus > wrote: >> The path isn't dynamically generated based on the domain, it's stored per-file in the database. >> You can update the db or symlink the folders so they are available in the old structure too. >> -Avi >> >> On Mon, Nov 17, 2014 at 10:35 AM, Joel White > wrote: >> I initially configured my FreeSWITCH with an IP address, but now for scalability and high availability I am changing to a domain name. I successfully changed the system over, but was wondering how do I move the voicemails to the new domain folder under voicemail? >> >> The obvious answer is to move the files, I already did that and restarted the system. The existing voicemails and greetings are not visible to FreeSWITCH when I call voicemail. >> >> How do I transfer them over? >> >> Thank you in advance for any light you may shed on this predicament -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/a9d22007/attachment.html From msc at freeswitch.org Tue Nov 18 00:33:25 2014 From: msc at freeswitch.org (Michael Collins) Date: Mon, 17 Nov 2014 13:33:25 -0800 Subject: [Freeswitch-users] lua script to disconnect active outbound calls to a specific gateway In-Reply-To: <88A6AB16FA7E42F88178F237CAF0D97B@bob> References: <88A6AB16FA7E42F88178F237CAF0D97B@bob> Message-ID: I'm sure it's possible. I think the bigger question is: what is the endgame? Is there a bigger picture here? In any case, if you're not using the presence_data field you could always set that value on the outbound leg and then find the corresponding channels with "show channels like 'foo'" and from there do something like "uuid_kill " on each one. In any case, look before you leap. Using a tool that supposedly kills all calls to a particular gateway may have unintended consequences. -MC On Mon, Nov 17, 2014 at 8:11 AM, Jason Holden wrote: > Is this possible? > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/dd9cb47a/attachment.html From brian at freeswitch.org Tue Nov 18 01:40:41 2014 From: brian at freeswitch.org (Brian West) Date: Mon, 17 Nov 2014 16:40:41 -0600 Subject: [Freeswitch-users] lua script to disconnect active outbound calls to a specific gateway In-Reply-To: References: <88A6AB16FA7E42F88178F237CAF0D97B@bob> Message-ID: fsctl hupall matching var Been there since 2010 :) On Mon, Nov 17, 2014 at 3:33 PM, Michael Collins wrote: > I'm sure it's possible. I think the bigger question is: what is the > endgame? Is there a bigger picture here? > > In any case, if you're not using the presence_data field you could always > set that value on the outbound leg and then find the corresponding channels > with "show channels like 'foo'" and from there do something like "uuid_kill > " on each one. > > In any case, look before you leap. Using a tool that supposedly kills all > calls to a particular gateway may have unintended consequences. > > -MC > > > On Mon, Nov 17, 2014 at 8:11 AM, Jason Holden > wrote: > >> Is this possible? >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/d7177c14/attachment-0001.html From steveayre at gmail.com Tue Nov 18 02:36:42 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 17 Nov 2014 23:36:42 +0000 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: <5F2EF027-BFA2-4ACF-A08A-12D50083DED6@jerris.com> References: <5F2EF027-BFA2-4ACF-A08A-12D50083DED6@jerris.com> Message-ID: Ah ok. Not sure what I was thinking then actually as we're on 14.04 not 12.04. So Bruce my original comment stands, best to use the latest LTS 'Trusty' On 17 November 2014 14:25, Michael Jerris wrote: > Random crashes. We have traced a number of them down to changes/patches > pulled in to system libs, but stopped bothering, its not worth the time as > the new LTS works fine. > > > On Nov 17, 2014, at 9:01 AM, Steven Ayre wrote: > > That being said, we have had a string of well known very bad behaviors due >> to bugs in 12.04 > > > Can you elaborate Michael? (The company I work for runs FS exclusively on > servers running 12.04, although my personal preference would be Debian). > > On 17 November 2014 13:46, Michael Jerris wrote: > >> Build on 12.04 generally works if you fix all the dependency issues >> (there are several). That being said, we have had a string of well known >> very bad behaviors due to bugs in 12.04. We strongly recommend against >> using it. You will have random crashes and we will not spend any time >> trying to fix them due to these known issues. >> >> >> On Nov 15, 2014, at 5:58 AM, Steven Ayre wrote: >> >> I believe 12.04 is too old to be supported by FreeSWITCH, it might be >> worth upgrading to 14.04 Trusty (which is also a LTS release) and which I >> know FS builds fine on on. >> >> On 14 November 2014 21:47, Bruce Lefko wrote: >> >>> So I ran those two exports you gave me and added "-c precise" to my >>> utils call, but it still seems to be running into that pbuilder bug. >>> Here's the output or log/precise-amd64: >>> >>> pbuilder/build//cow.20008 cow-shell >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Upgrading for distribution precise >>> I: copying local configuration >>> cp: cannot create regular file >>> `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory >>> pbuilder update failed >>> E: could not update with cowdancer, try --no-cowdancer-update option >>> forking: rm -rf /var/cache/pbuilder/build//cow.20008 >>> -> Copying COW directory >>> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >>> forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow >>> /var/cache/pbuilder/build//cow.22359 >>> I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist >>> failed, it didn't exist? >>> -> Invoking pbuilder >>> forking: pbuilder update --override-config --buildplace >>> /var/cache/pbuilder/build//cow.22359 --mirror >>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise >>> --no-targz --internal-chrootexec chroot >>> /var/cache/pbuilder/build//cow.22359 cow-shell >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Upgrading for distribution precise >>> I: copying local configuration >>> cp: cannot create regular file >>> `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory >>> pbuilder update failed >>> E: could not update with cowdancer, try --no-cowdancer-update option >>> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >>> cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, >>> does not exist >>> >>> Thanks! >>> >>> >>> >>> On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko >>> wrote: >>> >>>> Yeah, so it looks like in log/sid-amd64 I see the following over and >>>> over again: >>>> >>>> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >>>> -> Invoking pbuilder >>>> forking: pbuilder create --buildplace >>>> /var/cache/pbuilder/base-sid-amd64.cow --mirror >>>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >>>> --distribution sid --no-targz --extrapackages cowdancer >>>> W: /root/.pbuilderrc does not exist >>>> I: Running in no-targz mode >>>> I: Distribution is sid. >>>> I: Building the build environment >>>> I: running debootstrap >>>> /usr/sbin/debootstrap >>>> I: Retrieving Release >>>> E: Failed getting release file >>>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >>>> E: debootstrap failed >>>> W: Aborting with an error >>>> pbuilder create failed >>>> >>>> >>>> >>>> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >>>> wrote: >>>> >>>>> I am trying to tweak mod_spandsp in master and deb package the code >>>>> for use in my application. >>>>> >>>>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>>>> but it consistently gets stuck at "Building sid-amd64 debs.." >>>>> >>>>> First off, I'm not sure if this is the correct way to package custom >>>>> changes, but also why is the debian build stuck at this step? >>>>> >>>>> Thanks! >>>>> >>>> >>>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/cefc1098/attachment.html From robb at boardman.me.uk Tue Nov 18 02:19:45 2014 From: robb at boardman.me.uk (Robert Boardman) Date: Mon, 17 Nov 2014 23:19:45 +0000 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: Thanks for your replies, yes I need to run on win32 because someone bought the wrong windows version, how easy is it to convert the javascript to lua? shown below var exit = false; function onInput( session, type, data, arg ) { if ( type == "dtmf" ) { console_log( "info", "Got digit " + data.digit + "\n"); if ( data.digit == "5" ) { exit = true; return( false ); } else if ( data.digit == "#" ) { return( "seek:0" ); } else if ( data.digit == "1" ) { return( "seek:-500" ); } else if ( data.digit == "*" ) { return( "pause" ); } else if ( data.digit == "3" ) { return( "seek:+500" ); } return( true ); } } session = new Session('sofia/gateway/sipuser/789'); session.waitForAnswer(10000); if ( session.ready( ) ) { session.answer( ); while ( session.ready( ) && ! exit ) { session.streamFile( "c:/wamp/www/tmp/robb193497-tmp.wav", onInput ); session.hangup( ); } if ( session.ready( ) ) { session.hangup( ); } } Thanks Robb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/2de550ea/attachment.html From mike at jerris.com Tue Nov 18 06:04:55 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 17 Nov 2014 22:04:55 -0500 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: <5F2EF027-BFA2-4ACF-A08A-12D50083DED6@jerris.com> Message-ID: <5A2945B5-0A79-4EF5-9EF0-61AB53009EB3@jerris.com> If we are talking about "best" then I would say Debian 7 is the preferred platform as that is what we do the most testing and development on. > On Nov 17, 2014, at 6:36 PM, Steven Ayre wrote: > > Ah ok. Not sure what I was thinking then actually as we're on 14.04 not 12.04. So Bruce my original comment stands, best to use the latest LTS 'Trusty' > > On 17 November 2014 14:25, Michael Jerris > wrote: > Random crashes. We have traced a number of them down to changes/patches pulled in to system libs, but stopped bothering, its not worth the time as the new LTS works fine. > > >> On Nov 17, 2014, at 9:01 AM, Steven Ayre > wrote: >> >> That being said, we have had a string of well known very bad behaviors due to bugs in 12.04 >> >> Can you elaborate Michael? (The company I work for runs FS exclusively on servers running 12.04, although my personal preference would be Debian). >> >> On 17 November 2014 13:46, Michael Jerris > wrote: >> Build on 12.04 generally works if you fix all the dependency issues (there are several). That being said, we have had a string of well known very bad behaviors due to bugs in 12.04. We strongly recommend against using it. You will have random crashes and we will not spend any time trying to fix them due to these known issues. >> >> >>> On Nov 15, 2014, at 5:58 AM, Steven Ayre > wrote: >>> >>> I believe 12.04 is too old to be supported by FreeSWITCH, it might be worth upgrading to 14.04 Trusty (which is also a LTS release) and which I know FS builds fine on on. >>> >>> On 14 November 2014 21:47, Bruce Lefko > wrote: >>> So I ran those two exports you gave me and added "-c precise" to my utils call, but it still seems to be running into that pbuilder bug. Here's the output or log/precise-amd64: >>> >>> pbuilder/build//cow.20008 cow-shell >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Upgrading for distribution precise >>> I: copying local configuration >>> cp: cannot create regular file `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory >>> pbuilder update failed >>> E: could not update with cowdancer, try --no-cowdancer-update option >>> forking: rm -rf /var/cache/pbuilder/build//cow.20008 >>> -> Copying COW directory >>> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >>> forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow /var/cache/pbuilder/build//cow.22359 >>> I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist failed, it didn't exist? >>> -> Invoking pbuilder >>> forking: pbuilder update --override-config --buildplace /var/cache/pbuilder/build//cow.22359 --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise --no-targz --internal-chrootexec chroot /var/cache/pbuilder/build//cow.22359 cow-shell >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Upgrading for distribution precise >>> I: copying local configuration >>> cp: cannot create regular file `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory >>> pbuilder update failed >>> E: could not update with cowdancer, try --no-cowdancer-update option >>> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >>> cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, does not exist >>> >>> Thanks! >>> >>> >>> >>> On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko > wrote: >>> Yeah, so it looks like in log/sid-amd64 I see the following over and over again: >>> >>> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >>> -> Invoking pbuilder >>> forking: pbuilder create --buildplace /var/cache/pbuilder/base-sid-amd64.cow --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 --distribution sid --no-targz --extrapackages cowdancer >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Distribution is sid. >>> I: Building the build environment >>> I: running debootstrap >>> /usr/sbin/debootstrap >>> I: Retrieving Release >>> E: Failed getting release file http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >>> E: debootstrap failed >>> W: Aborting with an error >>> pbuilder create failed >>> >>> >>> >>> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko > wrote: >>> I am trying to tweak mod_spandsp in master and deb package the code for use in my application. >>> >>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" but it consistently gets stuck at "Building sid-amd64 debs.." >>> >>> First off, I'm not sure if this is the correct way to package custom changes, but also why is the debian build stuck at this step? >>> >>> Thanks! >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141117/0d07d843/attachment-0001.html From akhilgarg7 at gmail.com Tue Nov 18 08:39:43 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Tue, 18 Nov 2014 11:09:43 +0530 Subject: [Freeswitch-users] fs_cli hangs Message-ID: running "fs_cli -d 7" gives the following output: [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /root/.fs_cli_conf. [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is /etc/fs_cli.conf. [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin profile [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1465 esl_send() SEND auth ClueCon [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:1465 esl_send() SEND log Regards, Akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/afbe004b/attachment.html From peter at olssononline.se Tue Nov 18 10:33:58 2014 From: peter at olssononline.se (Peter Olsson) Date: Tue, 18 Nov 2014 08:33:58 +0100 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: If you build it yourself, it should be included when building for 32-bit. I'm not sure it's included in the pre-compiled packages though. 2014-11-18 0:19 GMT+01:00 Robert Boardman : > > > Thanks for your replies, yes I need to run on win32 because someone bought > the wrong windows version, how easy is it to convert the javascript to lua? > shown below > > var exit = false; > > function onInput( session, type, data, arg ) { > if ( type == "dtmf" ) { > console_log( "info", "Got digit " + data.digit + "\n"); > if ( data.digit == "5" ) { > exit = true; > return( false ); > > } > else if ( data.digit == "#" ) { > return( "seek:0" ); > > } > else if ( data.digit == "1" ) { > return( "seek:-500" ); > > } > else if ( data.digit == "*" ) { > return( "pause" ); > > } > else if ( data.digit == "3" ) { > return( "seek:+500" ); > > } > return( true ); > > } > > } > session = new Session('sofia/gateway/sipuser/789'); > session.waitForAnswer(10000); > > if ( session.ready( ) ) { > session.answer( ); > while ( session.ready( ) && ! exit ) { > session.streamFile( "c:/wamp/www/tmp/robb193497-tmp.wav", onInput ); > session.hangup( ); > } > if ( session.ready( ) ) { > session.hangup( ); > > } > > } > > Thanks > Robb > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/54a0a641/attachment.html From pnlsantos at gmail.com Tue Nov 18 12:57:09 2014 From: pnlsantos at gmail.com (Pedro Santos) Date: Tue, 18 Nov 2014 09:57:09 +0000 Subject: [Freeswitch-users] perl esl In-Reply-To: References: Message-ID: I already done that. didn't resolve On Mon, Nov 17, 2014 at 6:39 PM, Darren Wiebe wrote: > Brian helped on IRC on Saturday with the same question for Debian. > > On Centos try "yum install perl-devel" > > Darren Wiebe > > On Mon, Nov 17, 2014 at 11:08 AM, Pedro Santos > wrote: > >> Hi, >> >> i can not install the perlmod esl. >> There is no perl/ESL.so file. >> i am using centos 6 and the stable version from git of freeswitch. >> >> Regards >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/c88374c7/attachment.html From mike at jerris.com Tue Nov 18 16:01:55 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 18 Nov 2014 08:01:55 -0500 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: if it builds fine then we should just add it to the packages, please open a jira On Tuesday, November 18, 2014, Peter Olsson wrote: > If you build it yourself, it should be included when building for 32-bit. > I'm not sure it's included in the pre-compiled packages though. > > 2014-11-18 0:19 GMT+01:00 Robert Boardman >: > >> >> >> Thanks for your replies, yes I need to run on win32 because someone >> bought the wrong windows version, how easy is it to convert the javascript >> to lua? shown below >> >> var exit = false; >> >> function onInput( session, type, data, arg ) { >> if ( type == "dtmf" ) { >> console_log( "info", "Got digit " + data.digit + "\n"); >> if ( data.digit == "5" ) { >> exit = true; >> return( false ); >> >> } >> else if ( data.digit == "#" ) { >> return( "seek:0" ); >> >> } >> else if ( data.digit == "1" ) { >> return( "seek:-500" ); >> >> } >> else if ( data.digit == "*" ) { >> return( "pause" ); >> >> } >> else if ( data.digit == "3" ) { >> return( "seek:+500" ); >> >> } >> return( true ); >> >> } >> >> } >> session = new Session('sofia/gateway/sipuser/789'); >> session.waitForAnswer(10000); >> >> if ( session.ready( ) ) { >> session.answer( ); >> while ( session.ready( ) && ! exit ) { >> session.streamFile( "c:/wamp/www/tmp/robb193497-tmp.wav", onInput ); >> session.hangup( ); >> } >> if ( session.ready( ) ) { >> session.hangup( ); >> >> } >> >> } >> >> Thanks >> Robb >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/05a1407e/attachment-0001.html From mike at jerris.com Tue Nov 18 16:02:52 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 18 Nov 2014 08:02:52 -0500 Subject: [Freeswitch-users] perl esl In-Reply-To: References: Message-ID: you need to run configure again after installing additional dependencies On Tuesday, November 18, 2014, Pedro Santos wrote: > I already done that. > didn't resolve > > On Mon, Nov 17, 2014 at 6:39 PM, Darren Wiebe > wrote: > >> Brian helped on IRC on Saturday with the same question for Debian. >> >> On Centos try "yum install perl-devel" >> >> Darren Wiebe >> >> On Mon, Nov 17, 2014 at 11:08 AM, Pedro Santos > > wrote: >> >>> Hi, >>> >>> i can not install the perlmod esl. >>> There is no perl/ESL.so file. >>> i am using centos 6 and the stable version from git of freeswitch. >>> >>> Regards >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/43b52bac/attachment.html From dgarcia at anew.com.ve Tue Nov 18 16:39:43 2014 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 18 Nov 2014 09:09:43 -0430 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: <546B4C1F.6000603@anew.com.ve> Hi, Have your tried contact Microsoft Support directly and ask if you can use your serial key to install a windows x64 bit instead of Win32. I got a case with Windows 7, the customer bought the wrong windows version for their workstations, they ask to Microsfot what they can do. At the end they just got a dvd installer for windows 64 bit and used the previous windwows key without problem. The key was valid for both and only the "package model" was the same, example, the same serial was valid for windows standard 32 o x64 but it become valid for windows enterprise. On 11/17/2014 06:49 PM, Robert Boardman wrote: > > > Thanks for your replies, yes I need to run on win32 because someone > bought the wrong windows version, how easy is it to convert the > javascript to lua? shown below > > var exit = false; > > function onInput( session, type, data, arg ) { > if ( type == "dtmf" ) { > console_log( "info", "Got digit " + data.digit + "\n"); > if ( data.digit == "5" ) { > exit = true; > return( false ); > > } > else if ( data.digit == "#" ) { > return( "seek:0" ); > > } > else if ( data.digit == "1" ) { > return( "seek:-500" ); > > } > else if ( data.digit == "*" ) { > return( "pause" ); > > } > else if ( data.digit == "3" ) { > return( "seek:+500" ); > > } > return( true ); > > } > > } > session = new Session('sofia/gateway/sipuser/789'); > session.waitForAnswer(10000); > > if ( session.ready( ) ) { > session.answer( ); > while ( session.ready( ) && ! exit ) { > session.streamFile( "c:/wamp/www/tmp/robb193497-tmp.wav", onInput ); > session.hangup( ); > } > if ( session.ready( ) ) { > session.hangup( ); > > } > > } > > Thanks > Robb > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/f2cf77cb/attachment.html From joelewhite at gmail.com Tue Nov 18 16:58:54 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 18 Nov 2014 08:58:54 -0500 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> References: <546A4215.2090509@earthspike.net> <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> Message-ID: Thank you Changing the domain worked. Here is my file path as it has existed /usr/local/freeswitch/storage/voicemail/default/ 10.111.252.10/26343/msg_418e9e26-df74-11e3-b0cd-d9415cbdfb71.wav I just need the IP changed to the domain I will try the update of the file path today On Mon, Nov 17, 2014 at 1:56 PM, Michael Jerris wrote: > This does seem kind of clumsy... I'd be happy to accept a patch for > mod_voicemail change domain that moves files and updates db correctly if > anyone wants to give it a try. > > On Nov 17, 2014, at 1:44 PM, John wrote: > > I've done this and for exactly the same reason. I moved the files to the > domain-named path and updated the database using sqlite3. I found this in > my command history, but you should test first on a copy of the db. It's for > upgrading from 1.2.quite_old to 1.4.quite_recent. > > $ sqlite3 voicemail_default.db > sqlite> update voicemail_msgs set file_path='/var/lib' || > substr(replace(file_path,'172.16.12.34','pbx.somewhere.net'),11); > sqlite> update voicemail_msgs set > file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net'); > sqlite> update voicemail_prefs set greeting_path='/var/lib' || > substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net'),11); > > I cannot remember what the '11' was for; I've changed the IP address and > FQDN so you might need to tune this value; check with SELECTs in place of > the UPDATEs. > > John > > On 17/11/14 18:08, Ken Rice wrote: > > You got it sqlite3 is the way to go... I doubt doing what you are trying > is heavily documented tho as its not something normally done. > > But all metadata for voicemail is in the sqlite db and that includes file > paths etc > > > On 11/17/14 10:07 AM, "Joel White" wrote: > > What is the easiest way to make this change? > > SQLite3 via command line? Or is there another way to manipulate the > database? > > On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: > > The path isn't dynamically generated based on the domain, it's stored > per-file in the database. > You can update the db or symlink the folders so they are available in the > old structure too. > -Avi > > On Mon, Nov 17, 2014 at 10:35 AM, Joel White wrote: > > I initially configured my FreeSWITCH with an IP address, but now for > scalability and high availability I am changing to a domain name. I > successfully changed the system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did that and restarted > the system. The existing voicemails and greetings are not visible to > FreeSWITCH when I call voicemail. > > How do I transfer them over? > > Thank you in advance for any light you may shed on this predicament > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/65cca10d/attachment-0001.html From joelewhite at gmail.com Tue Nov 18 17:21:25 2014 From: joelewhite at gmail.com (Joel White) Date: Tue, 18 Nov 2014 09:21:25 -0500 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: References: <546A4215.2090509@earthspike.net> <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> Message-ID: The change went smoothly. I really only used the update ***=replace function to change out the IP to Domain Everything looks good On Tue, Nov 18, 2014 at 8:58 AM, Joel White wrote: > Thank you > > Changing the domain worked. > > Here is my file path as it has existed > > /usr/local/freeswitch/storage/voicemail/default/ > 10.111.252.10/26343/msg_418e9e26-df74-11e3-b0cd-d9415cbdfb71.wav > > I just need the IP changed to the domain > > I will try the update of the file path today > > On Mon, Nov 17, 2014 at 1:56 PM, Michael Jerris wrote: > >> This does seem kind of clumsy... I'd be happy to accept a patch for >> mod_voicemail change domain that moves files and updates db correctly if >> anyone wants to give it a try. >> >> On Nov 17, 2014, at 1:44 PM, John wrote: >> >> I've done this and for exactly the same reason. I moved the files to the >> domain-named path and updated the database using sqlite3. I found this in >> my command history, but you should test first on a copy of the db. It's for >> upgrading from 1.2.quite_old to 1.4.quite_recent. >> >> $ sqlite3 voicemail_default.db >> sqlite> update voicemail_msgs set file_path='/var/lib' || >> substr(replace(file_path,'172.16.12.34','pbx.somewhere.net'),11); >> sqlite> update voicemail_msgs set >> file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net'); >> sqlite> update voicemail_prefs set greeting_path='/var/lib' || >> substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net'),11); >> >> I cannot remember what the '11' was for; I've changed the IP address and >> FQDN so you might need to tune this value; check with SELECTs in place of >> the UPDATEs. >> >> John >> >> On 17/11/14 18:08, Ken Rice wrote: >> >> You got it sqlite3 is the way to go... I doubt doing what you are trying >> is heavily documented tho as its not something normally done. >> >> But all metadata for voicemail is in the sqlite db and that includes file >> paths etc >> >> >> On 11/17/14 10:07 AM, "Joel White" wrote: >> >> What is the easiest way to make this change? >> >> SQLite3 via command line? Or is there another way to manipulate the >> database? >> >> On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: >> >> The path isn't dynamically generated based on the domain, it's stored >> per-file in the database. >> You can update the db or symlink the folders so they are available in the >> old structure too. >> -Avi >> >> On Mon, Nov 17, 2014 at 10:35 AM, Joel White >> wrote: >> >> I initially configured my FreeSWITCH with an IP address, but now for >> scalability and high availability I am changing to a domain name. I >> successfully changed the system over, but was wondering how do I move the >> voicemails to the new domain folder under voicemail? >> >> The obvious answer is to move the files, I already did that and restarted >> the system. The existing voicemails and greetings are not visible to >> FreeSWITCH when I call voicemail. >> >> How do I transfer them over? >> >> Thank you in advance for any light you may shed on this predicament >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/4a2595e5/attachment.html From brian at freeswitch.org Tue Nov 18 19:38:05 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Nov 2014 10:38:05 -0600 Subject: [Freeswitch-users] Javascript scripting In-Reply-To: References: Message-ID: see scripts/lua/callback.lua On Mon, Nov 17, 2014 at 5:19 PM, Robert Boardman wrote: > > > Thanks for your replies, yes I need to run on win32 because someone bought > the wrong windows version, how easy is it to convert the javascript to lua? > shown below > > var exit = false; > > function onInput( session, type, data, arg ) { > if ( type == "dtmf" ) { > console_log( "info", "Got digit " + data.digit + "\n"); > if ( data.digit == "5" ) { > exit = true; > return( false ); > > } > else if ( data.digit == "#" ) { > return( "seek:0" ); > > } > else if ( data.digit == "1" ) { > return( "seek:-500" ); > > } > else if ( data.digit == "*" ) { > return( "pause" ); > > } > else if ( data.digit == "3" ) { > return( "seek:+500" ); > > } > return( true ); > > } > > } > session = new Session('sofia/gateway/sipuser/789'); > session.waitForAnswer(10000); > > if ( session.ready( ) ) { > session.answer( ); > while ( session.ready( ) && ! exit ) { > session.streamFile( "c:/wamp/www/tmp/robb193497-tmp.wav", onInput ); > session.hangup( ); > } > if ( session.ready( ) ) { > session.hangup( ); > > } > > } > > Thanks > Robb > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/6e06f489/attachment-0001.html From brian at freeswitch.org Tue Nov 18 19:39:25 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Nov 2014 10:39:25 -0600 Subject: [Freeswitch-users] fs_cli hangs In-Reply-To: References: Message-ID: Does starting FreeSWITCH without any args also hang? On Mon, Nov 17, 2014 at 11:39 PM, akhil garg wrote: > running "fs_cli -d 7" gives the following output: > > > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /root/.fs_cli_conf. > [DEBUG] esl_config.c:56 esl_config_open_file() Configuration file is > /etc/fs_cli.conf. > [DEBUG] fs_cli.c:1438 main() profile default does not exist using builtin > profile > [DEBUG] fs_cli.c:1468 main() Using profile internal [127.0.0.1] > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = > [auth/request] > [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE > Event-Name: SOCKET_DATA > Content-Type: auth/request > > > [DEBUG] esl.c:1465 esl_send() SEND > auth ClueCon > > > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Content-Type] = > [command/reply] > [DEBUG] esl.c:1265 esl_recv_event() RECV HEADER [Reply-Text] = [+OK > accepted] > [DEBUG] esl.c:1437 esl_recv_event() RECV MESSAGE > Event-Name: SOCKET_DATA > Content-Type: command/reply > Reply-Text: +OK accepted > > > [DEBUG] esl.c:1465 esl_send() SEND > log > > > > > > Regards, > Akhil > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/313bbe8a/attachment.html From leon.ramos at tiendaip.mx Tue Nov 18 20:36:10 2014 From: leon.ramos at tiendaip.mx (leon.ramos at tiendaip.mx) Date: Tue, 18 Nov 2014 11:36:10 -0600 Subject: [Freeswitch-users] freeswitch + openr2 In-Reply-To: References: <20141029013738.10136t0hse0z8182@webmail.tiendaip.mx> Message-ID: <20141118113610.20173rbafua8tfgq@webmail.tiendaip.mx> Hi Moises! Everything ok over here, what about the cold over there? And thanks, I did what you suggested. The code from github worked pretty easy, I have just ajusted some variables in my spec file and now I have a new rpm based on your latest change on 24 may 2012. I am rebuilding now the ISO for quick install. I was sharing the ISO in sourceforge, but what do you suggest me to use nowadays? Saludos! Moises Silva escribi?: > Hi Leon, hope things are going good for you in Mexico :) > > On Wed, Oct 29, 2014 at 2:37 AM, wrote: > >> Dear list: >> >> I am facing a problem when trying to compile freeswitch, freetdm + openr2. >> >> I followed the wiki on how to do it, I was able to use the old 1.2 >> openr2 version. >> >> A. If I use the version 1.2 from the svn trunk and the cmake command >> > > You should use github master: https://github.com/moises-silva/openr2 > > I can't recall if there's differences though, but svn is definitely not the > best option now days, I might have done changes in git that are not in svn > anymore (I'm surprised svn still works, I thought I had removed it) > > Try using master from github, "cd build/ && cmake .. && make install" or > something like that > > Then when freetdm is configured, if it still does not detect openr2, please > share config.log using pastebin (it's located in the freetdm/ directory, is > a file produced when running ./configure) > > Moy > > Moises Silva > > Manager, Software Engineering > > > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, Ontario, Canada L3R 9R6 > > T +1 905 474 1990 x128 | toll-free in North America +1 800 388-2475 | F +1 > 905 474 9223 > > www.sangoma.com > From covici at ccs.covici.com Tue Nov 18 22:09:27 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 18 Nov 2014 14:09:27 -0500 Subject: [Freeswitch-users] segfault on git from the 17th Message-ID: <27964.1416337767@ccs.covici.com> Hi. I am getting segfaults when some calls are destroyed, but I can't reproduce it on demand. Should I file it anyway, even though I can't answer the question about latest git or how should I do this? The last few lines of the stack trace are: f348c00d020, obj=) at src/switch_core_session.c:1690 #26 0x00007f3495540204 in start_thread (arg=0x7f343c169700) at pthread_create.c:310 #27 0x00007f3494c4d90d in clone () at ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From krice at freeswitch.org Tue Nov 18 22:14:04 2014 From: krice at freeswitch.org (Ken Rice) Date: Tue, 18 Nov 2014 13:14:04 -0600 Subject: [Freeswitch-users] segfault on git from the 17th In-Reply-To: <27964.1416337767@ccs.covici.com> Message-ID: A full stack trace is needed to determine anything On 11/18/14 1:09 PM, "covici at ccs.covici.com" wrote: > Hi. I am getting segfaults when some calls are destroyed, but I can't > reproduce it on demand. Should I file it anyway, even though I can't > answer the question about latest git or how should I do this? > The last few lines of the stack trace are: > > f348c00d020, obj=) at src/switch_core_session.c:1690 > #26 0x00007f3495540204 in start_thread (arg=0x7f343c169700) at > pthread_create.c:310 > #27 0x00007f3494c4d90d in clone () at > ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 > > Thanks. > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH From brian at freeswitch.org Tue Nov 18 22:15:41 2014 From: brian at freeswitch.org (Brian West) Date: Tue, 18 Nov 2014 13:15:41 -0600 Subject: [Freeswitch-users] segfault on git from the 17th In-Reply-To: <27964.1416337767@ccs.covici.com> References: <27964.1416337767@ccs.covici.com> Message-ID: Could have been FS-7014, try master we fixed that segfault this morning. On Tue, Nov 18, 2014 at 1:09 PM, wrote: > Hi. I am getting segfaults when some calls are destroyed, but I can't > reproduce it on demand. Should I file it anyway, even though I can't > answer the question about latest git or how should I do this? > The last few lines of the stack trace are: > > f348c00d020, obj=) at src/switch_core_session.c:1690 > #26 0x00007f3495540204 in start_thread (arg=0x7f343c169700) at > pthread_create.c:310 > #27 0x00007f3494c4d90d in clone () at > ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 > > Thanks. > > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/7f3f3bef/attachment.html From covici at ccs.covici.com Tue Nov 18 22:21:20 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 18 Nov 2014 14:21:20 -0500 Subject: [Freeswitch-users] segfault on git from the 17th In-Reply-To: References: Message-ID: <29720.1416338480@ccs.covici.com> I have that, but my question is about how to answer the latest git question. Ken Rice wrote: > A full stack trace is needed to determine anything > > > On 11/18/14 1:09 PM, "covici at ccs.covici.com" wrote: > > > Hi. I am getting segfaults when some calls are destroyed, but I can't > > reproduce it on demand. Should I file it anyway, even though I can't > > answer the question about latest git or how should I do this? > > The last few lines of the stack trace are: > > > > f348c00d020, obj=) at src/switch_core_session.c:1690 > > #26 0x00007f3495540204 in start_thread (arg=0x7f343c169700) at > > pthread_create.c:310 > > #27 0x00007f3494c4d90d in clone () at > > ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 > > > > Thanks. > > > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > Twitter: @FreeSWITCH > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From covici at ccs.covici.com Tue Nov 18 22:22:03 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 18 Nov 2014 14:22:03 -0500 Subject: [Freeswitch-users] segfault on git from the 17th In-Reply-To: References: <27964.1416337767@ccs.covici.com> Message-ID: <29828.1416338523@ccs.covici.com> This is what I have, so I will hold off and see if it happens again. Brian West wrote: > Could have been FS-7014, try master we fixed that segfault this morning. > > On Tue, Nov 18, 2014 at 1:09 PM, wrote: > > > Hi. I am getting segfaults when some calls are destroyed, but I can't > > reproduce it on demand. Should I file it anyway, even though I can't > > answer the question about latest git or how should I do this? > > The last few lines of the stack trace are: > > > > f348c00d020, obj=) at src/switch_core_session.c:1690 > > #26 0x00007f3495540204 in start_thread (arg=0x7f343c169700) at > > pthread_create.c:310 > > #27 0x00007f3494c4d90d in clone () at > > ../sysdeps/unix/sysv/linux/x86_64/clone.S:109 > > > > Thanks. > > > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici at ccs.covici.com > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From jason.holden at start.ca Tue Nov 18 23:21:46 2014 From: jason.holden at start.ca (Jason Holden) Date: Tue, 18 Nov 2014 15:21:46 -0500 Subject: [Freeswitch-users] lua script to disconnect active outboundcalls to a specific gateway In-Reply-To: References: <88A6AB16FA7E42F88178F237CAF0D97B@bob> Message-ID: <0010733A421F4420A1B51F0F92E38A0B@bob> I am looking to force disconnections of channels if for example an emergency number is dialed and there are active calls on a specific gateway where the emergency call would be routed to. The purpose of this is to ensure that the outbound emergency call connects to its end destination and does not experience any busy lines. From: Brian West [mailto:brian at freeswitch.org] Sent: Monday, November 17, 2014 5:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] lua script to disconnect active outboundcalls to a specific gateway fsctl hupall matching var Been there since 2010 :) On Mon, Nov 17, 2014 at 3:33 PM, Michael Collins wrote: I'm sure it's possible. I think the bigger question is: what is the endgame? Is there a bigger picture here? In any case, if you're not using the presence_data field you could always set that value on the outbound leg and then find the corresponding channels with "show channels like 'foo'" and from there do something like "uuid_kill " on each one. In any case, look before you leap. Using a tool that supposedly kills all calls to a particular gateway may have unintended consequences. -MC On Mon, Nov 17, 2014 at 8:11 AM, Jason Holden wrote: Is this possible? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/f55c2d1d/attachment.html From blefko5361 at gmail.com Wed Nov 19 00:19:30 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Tue, 18 Nov 2014 15:19:30 -0600 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: I've switched to 14.04, and my script with the commands I'm issuing is below, but I'm still getting the following in the logs: E: Package 'cowdancer' has no installation candidate I: unmounting dev/pts filesystem I: unmounting run/shm filesystem I: unmounting proc filesystem pbuilder create failed forking: rm -rf /var/cache/pbuilder/base-trusty-amd64.cow -> Invoking pbuilder forking: pbuilder create --buildplace /var/cache/pbuilder/base-trusty-amd64.cow --mirror http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 --distribution trusty --no-targz --extrapackages cowdancer Here's my script: #!/bin/bash apt-get update apt-get install -y git-core cd /usr/src git clone https://freeswitch.org/stash/scm/fs/freeswitch.git cd freeswitch cat >/tmp/modules.conf <<'EOL' applications/mod_commands applications/mod_conference applications/mod_db applications/mod_dptools applications/mod_enum applications/mod_esf applications/mod_esl applications/mod_expr applications/mod_fifo applications/mod_fsv applications/mod_hash applications/mod_httapi applications/mod_spandsp codecs/mod_amr codecs/mod_bv codecs/mod_g723_1 codecs/mod_g729 codecs/mod_h26x codecs/mod_vp8 codecs/mod_speex dialplans/mod_dialplan_asterisk dialplans/mod_dialplan_xml endpoints/mod_loopback endpoints/mod_sofia event_handlers/mod_cdr_csv event_handlers/mod_cdr_sqlite event_handlers/mod_event_socket formats/mod_local_stream formats/mod_native_file formats/mod_sndfile formats/mod_tone_stream languages/mod_lua languages/mod_spidermonkey loggers/mod_console loggers/mod_logfile loggers/mod_syslog say/mod_say_en xml_int/mod_xml_cdr xml_int/mod_xml_rpc xml_int/mod_xml_scgi EOL export COMPONENTS="main restricted universe multiverse" export NO_COWDANCER_UPDATE=1 ./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf -c trusty On Fri, Nov 14, 2014 at 3:47 PM, Bruce Lefko wrote: > So I ran those two exports you gave me and added "-c precise" to my utils > call, but it still seems to be running into that pbuilder bug. Here's the > output or log/precise-amd64: > > pbuilder/build//cow.20008 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file > `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.20008 > -> Copying COW directory > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow > /var/cache/pbuilder/build//cow.22359 > I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist > failed, it didn't exist? > -> Invoking pbuilder > forking: pbuilder update --override-config --buildplace > /var/cache/pbuilder/build//cow.22359 --mirror > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise > --no-targz --internal-chrootexec chroot > /var/cache/pbuilder/build//cow.22359 cow-shell > W: /root/.pbuilderrc does not exist > I: Running in no-targz mode > I: Upgrading for distribution precise > I: copying local configuration > cp: cannot create regular file > `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory > pbuilder update failed > E: could not update with cowdancer, try --no-cowdancer-update option > forking: rm -rf /var/cache/pbuilder/build//cow.22359 > cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, > does not exist > > Thanks! > > > > On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko wrote: > >> Yeah, so it looks like in log/sid-amd64 I see the following over and over >> again: >> >> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >> -> Invoking pbuilder >> forking: pbuilder create --buildplace >> /var/cache/pbuilder/base-sid-amd64.cow --mirror >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >> --distribution sid --no-targz --extrapackages cowdancer >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Distribution is sid. >> I: Building the build environment >> I: running debootstrap >> /usr/sbin/debootstrap >> I: Retrieving Release >> E: Failed getting release file >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >> E: debootstrap failed >> W: Aborting with an error >> pbuilder create failed >> >> >> >> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >> wrote: >> >>> I am trying to tweak mod_spandsp in master and deb package the code for >>> use in my application. >>> >>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>> but it consistently gets stuck at "Building sid-amd64 debs.." >>> >>> First off, I'm not sure if this is the correct way to package custom >>> changes, but also why is the debian build stuck at this step? >>> >>> Thanks! >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/6703e177/attachment-0001.html From freeswitch at earthspike.net Wed Nov 19 01:13:33 2014 From: freeswitch at earthspike.net (John) Date: Tue, 18 Nov 2014 22:13:33 +0000 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: References: <546A4215.2090509@earthspike.net> <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> Message-ID: <546BC48D.3020007@earthspike.net> I've just remembered why there were 2 updates of file_path. I moved the voicemails from a box compiled from git into /usr/local/freeswitch to an Ubuntu box with debian-style file paths. And I put a typo in the first update, so had to do the IP-to-FQDN replace as a second transaction. Anyway, glad you got it work despite my slightly misleading guidance! John On 18/11/14 14:21, Joel White wrote: > The change went smoothly. I really only used the update ***=replace > function to change out the IP to Domain > > Everything looks good > > On Tue, Nov 18, 2014 at 8:58 AM, Joel White > wrote: > > Thank you > > Changing the domain worked. > > Here is my file path as it has existed > > /usr/local/freeswitch/storage/voicemail/default/10.111.252.10/26343/msg_418e9e26-df74-11e3-b0cd-d9415cbdfb71.wav > > > I just need the IP changed to the domain > > I will try the update of the file path today > > On Mon, Nov 17, 2014 at 1:56 PM, Michael Jerris > wrote: > > This does seem kind of clumsy... I'd be happy to accept a > patch for mod_voicemail change domain that moves files and > updates db correctly if anyone wants to give it a try. > >> On Nov 17, 2014, at 1:44 PM, John > > wrote: >> >> I've done this and for exactly the same reason. I moved the >> files to the domain-named path and updated the database using >> sqlite3. I found this in my command history, but you should >> test first on a copy of the db. It's for upgrading from >> 1.2.quite_old to 1.4.quite_recent. >> >> $ sqlite3 voicemail_default.db >> sqlite> update voicemail_msgs set file_path='/var/lib' || >> substr(replace(file_path,'172.16.12.34','pbx.somewhere.net >> '),11); >> sqlite> update voicemail_msgs set >> file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net >> '); >> sqlite> update voicemail_prefs set greeting_path='/var/lib' >> || >> substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net '),11); >> >> I cannot remember what the '11' was for; I've changed the IP >> address and FQDN so you might need to tune this value; check >> with SELECTs in place of the UPDATEs. >> >> John >> >> On 17/11/14 18:08, Ken Rice wrote: >>> You got it sqlite3 is the way to go... I doubt doing what >>> you are trying is heavily documented tho as its not >>> something normally done. >>> >>> But all metadata for voicemail is in the sqlite db and that >>> includes file paths etc >>> >>> >>> On 11/17/14 10:07 AM, "Joel White" wrote: >>> >>> What is the easiest way to make this change? >>> >>> SQLite3 via command line? Or is there another way to >>> manipulate the database? >>> >>> On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus >>> wrote: >>> >>> The path isn't dynamically generated based on the >>> domain, it's stored per-file in the database. >>> You can update the db or symlink the folders so they >>> are available in the old structure too. >>> -Avi >>> >>> On Mon, Nov 17, 2014 at 10:35 AM, Joel White >>> wrote: >>> >>> I initially configured my FreeSWITCH with an IP >>> address, but now for scalability and high >>> availability I am changing to a domain name. I >>> successfully changed the system over, but was >>> wondering how do I move the voicemails to the >>> new domain folder under voicemail? >>> >>> The obvious answer is to move the files, I >>> already did that and restarted the system. The >>> existing voicemails and greetings are not >>> visible to FreeSWITCH when I call voicemail. >>> >>> How do I transfer them over? >>> >>> Thank you in advance for any light you may shed >>> on this predicament >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141118/de15f93e/attachment-0001.html From nneul at mst.edu Wed Nov 19 01:24:23 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 18 Nov 2014 16:24:23 -0600 Subject: [Freeswitch-users] feature idea for mod_callcenter Message-ID: <546BC717.8040906@mst.edu> Could be useful to have a "callcenter_config queue send [queue_name] [uuid] [agent]" that would explicitly send a call in the queue to a given agent, regardless of that agent's state. (Well, at least assuming they are capable of receiving a call.) The idea would be for when you have a "senior" agent that is in the system to take calls that are send to them from other agents or from a control panel, or as a 'agent handoff' to an agent that hasn't yet made themselves active in the queue. I can use just a plain transfer, but that requires jumping through extra hoops to change the state of the target agent - to make sure that they don't receive an actual queue call while in the middle of a transferred call that is no longer managed by mod_callcenter. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From kamil.nigmatullin at gmail.com Wed Nov 19 14:14:30 2014 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Wed, 19 Nov 2014 15:14:30 +0400 Subject: [Freeswitch-users] Presence question Message-ID: One of our clients wants to switch on presence feature, but they have 140 lines of FXS that does not support Publish/Subscribe function. Is there a way to achive this using data in freeswitch (if Registered or Busy)? I know that it is a silly question but they asked me to ask anyway, so will anybody just confirm that it is not possible? -- Kamil Nigmatullin Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141119/e00ac64c/attachment.html From blefko5361 at gmail.com Wed Nov 19 18:09:01 2014 From: blefko5361 at gmail.com (Bruce Lefko) Date: Wed, 19 Nov 2014 09:09:01 -0600 Subject: [Freeswitch-users] debian build-all freezing at "Building sid-amd64 debs" In-Reply-To: References: Message-ID: Can anyone confirm a reliable way to build FS 1.4 deb packages on ubuntu? I'll be happy to add the procedure to the documentation once I get something to work. Thanks! On Tue, Nov 18, 2014 at 3:19 PM, Bruce Lefko wrote: > I've switched to 14.04, and my script with the commands I'm issuing is > below, but I'm still getting the following in the logs: > E: Package 'cowdancer' has no installation candidate > I: unmounting dev/pts filesystem > I: unmounting run/shm filesystem > I: unmounting proc filesystem > pbuilder create failed > forking: rm -rf /var/cache/pbuilder/base-trusty-amd64.cow > -> Invoking pbuilder > forking: pbuilder create --buildplace > /var/cache/pbuilder/base-trusty-amd64.cow --mirror > http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 > --distribution trusty --no-targz --extrapackages cowdancer > > > Here's my script: > > #!/bin/bash > apt-get update > apt-get install -y git-core > cd /usr/src > > git clone https://freeswitch.org/stash/scm/fs/freeswitch.git > cd freeswitch > > cat >/tmp/modules.conf <<'EOL' > applications/mod_commands > applications/mod_conference > applications/mod_db > applications/mod_dptools > applications/mod_enum > applications/mod_esf > applications/mod_esl > applications/mod_expr > applications/mod_fifo > applications/mod_fsv > applications/mod_hash > applications/mod_httapi > applications/mod_spandsp > codecs/mod_amr > codecs/mod_bv > codecs/mod_g723_1 > codecs/mod_g729 > codecs/mod_h26x > codecs/mod_vp8 > codecs/mod_speex > dialplans/mod_dialplan_asterisk > dialplans/mod_dialplan_xml > endpoints/mod_loopback > endpoints/mod_sofia > event_handlers/mod_cdr_csv > event_handlers/mod_cdr_sqlite > event_handlers/mod_event_socket > formats/mod_local_stream > formats/mod_native_file > formats/mod_sndfile > formats/mod_tone_stream > languages/mod_lua > languages/mod_spidermonkey > loggers/mod_console > loggers/mod_logfile > loggers/mod_syslog > say/mod_say_en > xml_int/mod_xml_cdr > xml_int/mod_xml_rpc > xml_int/mod_xml_scgi > EOL > > export COMPONENTS="main restricted universe multiverse" > export NO_COWDANCER_UPDATE=1 > ./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf -c trusty > > On Fri, Nov 14, 2014 at 3:47 PM, Bruce Lefko wrote: > >> So I ran those two exports you gave me and added "-c precise" to my utils >> call, but it still seems to be running into that pbuilder bug. Here's the >> output or log/precise-amd64: >> >> pbuilder/build//cow.20008 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file >> `/var/cache/pbuilder/build/cow.20008/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.20008 >> -> Copying COW directory >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> forking: cp -al /var/cache/pbuilder/base-precise-amd64.cow >> /var/cache/pbuilder/build//cow.22359 >> I: unlink for ilistfile /var/cache/pbuilder/build//cow.22359/.ilist >> failed, it didn't exist? >> -> Invoking pbuilder >> forking: pbuilder update --override-config --buildplace >> /var/cache/pbuilder/build//cow.22359 --mirror >> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --distribution precise >> --no-targz --internal-chrootexec chroot >> /var/cache/pbuilder/build//cow.22359 cow-shell >> W: /root/.pbuilderrc does not exist >> I: Running in no-targz mode >> I: Upgrading for distribution precise >> I: copying local configuration >> cp: cannot create regular file >> `/var/cache/pbuilder/build/cow.22359/etc/hosts': No such file or directory >> pbuilder update failed >> E: could not update with cowdancer, try --no-cowdancer-update option >> forking: rm -rf /var/cache/pbuilder/build//cow.22359 >> cannot canonicalize filename /var/cache/pbuilder/base-precise-amd64.cow, >> does not exist >> >> Thanks! >> >> >> >> On Thu, Nov 13, 2014 at 7:47 PM, Bruce Lefko >> wrote: >> >>> Yeah, so it looks like in log/sid-amd64 I see the following over and >>> over again: >>> >>> forking: rm -rf /var/cache/pbuilder/base-sid-amd64.cow >>> -> Invoking pbuilder >>> forking: pbuilder create --buildplace >>> /var/cache/pbuilder/base-sid-amd64.cow --mirror >>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/ --architecture amd64 >>> --distribution sid --no-targz --extrapackages cowdancer >>> W: /root/.pbuilderrc does not exist >>> I: Running in no-targz mode >>> I: Distribution is sid. >>> I: Building the build environment >>> I: running debootstrap >>> /usr/sbin/debootstrap >>> I: Retrieving Release >>> E: Failed getting release file >>> http://us-west-2.ec2.archive.ubuntu.com/ubuntu/dists/sid/Release >>> E: debootstrap failed >>> W: Aborting with an error >>> pbuilder create failed >>> >>> >>> >>> On Thu, Nov 13, 2014 at 1:11 PM, Bruce Lefko >>> wrote: >>> >>>> I am trying to tweak mod_spandsp in master and deb package the code for >>>> use in my application. >>>> >>>> I'm running "./debian/util.sh build-all -ibn -z9 -f /tmp/modules.conf" >>>> but it consistently gets stuck at "Building sid-amd64 debs.." >>>> >>>> First off, I'm not sure if this is the correct way to package custom >>>> changes, but also why is the debian build stuck at this step? >>>> >>>> Thanks! >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141119/9ef1aae2/attachment.html From krice at freeswitch.org Wed Nov 19 18:19:03 2014 From: krice at freeswitch.org (Ken Rice) Date: Wed, 19 Nov 2014 09:19:03 -0600 Subject: [Freeswitch-users] ClueCon Weekly - Emil Ivov of Jitsi Message-ID: Don?t forget to join us at 1PM EST, 10AM PST, or 1800 GMT for ClueCon Weekly Today, Emil Ivov of Jitsi will be joining us. Find out what the jitsi guys have been working on! Join us at https://bbb.freeswitch.org/ and click the screen share button to see the screenshare/slides. (password is welcome) See you then! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141119/b2b26713/attachment.html From msc at freeswitch.org Wed Nov 19 20:31:17 2014 From: msc at freeswitch.org (Michael Collins) Date: Wed, 19 Nov 2014 09:31:17 -0800 Subject: [Freeswitch-users] lua script to disconnect active outbound calls to a specific gateway In-Reply-To: References: <88A6AB16FA7E42F88178F237CAF0D97B@bob> Message-ID: Thanks for the reminder. I totally forgot about that, as did anyone else who was going to get around to throwing that onto the wiki and/or add it to the fsctl hupall help text. -MC On Mon, Nov 17, 2014 at 2:40 PM, Brian West wrote: > fsctl hupall matching var > > Been there since 2010 :) > > On Mon, Nov 17, 2014 at 3:33 PM, Michael Collins > wrote: > >> I'm sure it's possible. I think the bigger question is: what is the >> endgame? Is there a bigger picture here? >> >> In any case, if you're not using the presence_data field you could always >> set that value on the outbound leg and then find the corresponding channels >> with "show channels like 'foo'" and from there do something like "uuid_kill >> " on each one. >> >> In any case, look before you leap. Using a tool that supposedly kills all >> calls to a particular gateway may have unintended consequences. >> >> -MC >> >> >> On Mon, Nov 17, 2014 at 8:11 AM, Jason Holden >> wrote: >> >>> Is this possible? >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141119/996f5075/attachment-0001.html From brian at freeswitch.org Wed Nov 19 23:24:03 2014 From: brian at freeswitch.org (Brian West) Date: Wed, 19 Nov 2014 14:24:03 -0600 Subject: [Freeswitch-users] lua script to disconnect active outboundcalls to a specific gateway In-Reply-To: <0010733A421F4420A1B51F0F92E38A0B@bob> References: <88A6AB16FA7E42F88178F237CAF0D97B@bob> <0010733A421F4420A1B51F0F92E38A0B@bob> Message-ID: gateway_name is set on every call that exits a gateway, uuid_dump a call to verify. On Tue, Nov 18, 2014 at 2:21 PM, Jason Holden wrote: > I am looking to force disconnections of channels if for example an > emergency number is dialed and there are active calls on a specific gateway > where the emergency call would be routed to. > > The purpose of this is to ensure that the outbound emergency call > connects to its end destination and does not experience any busy lines. > > > > > > *From:* Brian West [mailto:brian at freeswitch.org] > *Sent:* Monday, November 17, 2014 5:41 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] lua script to disconnect active > outboundcalls to a specific gateway > > > > fsctl hupall matching var > > Been there since 2010 :) > > > > On Mon, Nov 17, 2014 at 3:33 PM, Michael Collins > wrote: > > I'm sure it's possible. I think the bigger question is: what is the > endgame? Is there a bigger picture here? > > In any case, if you're not using the presence_data field you could always > set that value on the outbound leg and then find the corresponding channels > with "show channels like 'foo'" and from there do something like "uuid_kill > " on each one. > > In any case, look before you leap. Using a tool that supposedly kills all > calls to a particular gateway may have unintended consequences. > > -MC > > > > On Mon, Nov 17, 2014 at 8:11 AM, Jason Holden > wrote: > > Is this possible? > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141119/47d098d1/attachment.html From nghiaem at gmail.com Thu Nov 20 12:07:19 2014 From: nghiaem at gmail.com (=?UTF-8?B?Tmd1eeG7hW4gVsSDbiBOZ2jEqWEgRW0=?=) Date: Thu, 20 Nov 2014 16:07:19 +0700 Subject: [Freeswitch-users] Can't INVITE message to jitsi if length effective_caller_id_name >8 Message-ID: Hello, I have one problem of make call to sip exension which registered on jitsi. In dialplan if I set effective_caller_id_name a string value which its length > 8, the sip endpoint which use jitsi can't receive the call. But if I register that extension to other phones devices (AudioCodes 310HD, CSipSimple), these endpoint will receive the call (i used sofia_contact to ring multiple sip endpoint simultaneously). If the length of effective_caller_id_name <=8, jitsi sip endpoint will receive the call. I have capture sip messages on computer which install jitsi, I find that there are no INVITE message. But when caller cancel the call, I saw CANCEL message on this pc. I don't know that is the problem? Please help me. My FS version is 1.4.6 which run on RedHat 6 64 bits OS Thanks. -- Best regards, ------------------------------------------------------------------------------------------------------------- EM NGUYEN Mobile: +84 949669075 Yahoo & Skype: nghiaembt Website: http://freeswitch.vn -------------------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/86d9b0a6/attachment.html From david at kaymera.com Thu Nov 20 11:23:16 2014 From: david at kaymera.com (David Shemesh) Date: Thu, 20 Nov 2014 10:23:16 +0200 Subject: [Freeswitch-users] Opus Codec Error Message-ID: <546DA4F4.3040108@kaymera.com> Hope someone can help with this: I am experiencing a strange issue using Opus. The call is from a registered client to a landline through a sip Trunk. Freeswitch is transcoding From Opus to PCMU and after about ~1min the call drops with "switch_core_io.c:1282 Codec OPUS (STANDARD) decoder error!" and the call gets disconnected. It looks like this happens only when i enabling the jitterbuffer. I was able to reproduce this using various Clients (Csipsimple, Jitsi) & Sip trunks. it seems like its has something to do with freeswitch's implementation of OPUS. FreeSWITCH Version 1.5.15b+git~20141112T004420Z~dd629c1516~64bit Dialplan: Freeswitch Log: 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[opus:116:48000:20:0] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[PCMU:0:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3424 Audio Codec Compare [PCMU:0:8000:20:64000] is saved as a near-match 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[GSM:3:8000:20:13200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[PCMA:8:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[G722:9:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[AMR:96:8000:20:12200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[iLBC:97:8000:30:13330] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:8000:20:24600] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:16000:20:42200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:32000:20:44000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3315 Set telephone-event payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3494 Substituting codec PCMU at 10i@8000h 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:2343 Set Codec sofia/external/123456789 PCMU/8000 10 ms 80 samples 64000 bits 2014-11-18 10:51:25.876680 [DEBUG] switch_core_codec.c:111 sofia/external/123456789 Original read codec set to PCMU:0 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3623 Set 2833 dtmf send payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:4847 AUDIO RTP [sofia/external/123456789] 10.0.50.160 port 23644 -> 212.179.176.54 port 60036 codec: 0 ms: 10 2014-11-18 10:51:25.876680 [DEBUG] switch_rtp.c:3334 Starting timer [soft] 80 bytes per 10ms 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5194 Set 2833 dtmf send payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5200 Set 2833 dtmf receive payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5228 Set comfort noise payload to 13 2014-11-18 10:51:25.876680 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/123456789! 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3389 Send signal sofia/internal/1000 at 192.168.10.150 [BREAK] 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3393 (sofia/external/123456789) Callstate Change RINGING -> EARLY 2014-11-18 10:51:25.896682 [DEBUG] switch_core_codec.c:246 sofia/internal/1000 at 192.168.10.150 Restore previous codec opus:116. 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_originate.c:3551 Originate Resulted in Success: [sofia/external/123456789] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal sofia/internal/1000 at 192.168.10.150 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_bridge.c:1460 (sofia/external/123456789) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:1387 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:467 (sofia/external/123456789) Running State Change CS_EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:533 (sofia/external/123456789) State EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] mod_sofia.c:592 SOFIA EXCHANGE_MEDIA 2014-11-18 10:51:26.056685 [DEBUG] switch_rtp.c:5555 Correct ip/port confirmed. 2014-11-18 10:51:26.056685 [NOTICE] switch_core_io.c:1230 Activating write resampler 2014-11-18 10:51:26.056685 [DEBUG] switch_core_io.c:1458 Engaging Write Buffer at 160 bytes to accommodate 320->160 2014-11-18 10:51:26.136684 [ERR] switch_core_io.c:1282 Codec OPUS (STANDARD) decoder error! 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:578 sofia/external/123456789 ending bridge by request from write function 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD DONE [sofia/internal/1000 at 192.168.10.150] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD DONE [sofia/external/123456789] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal sofia/internal/1000 at 192.168.10.150 [BREAK] 2014-11-18 10:51:26.136684 [NOTICE] switch_ivr_bridge.c:751 Hangup sofia/external/123456789 [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-11-18 10:51:26.136684 [DEBUG] switch_channel.c:3215 Send signal sofia/external/123456789 [KILL] 2014-11-18 10:51:26.136684 [DEBUG] switch_core_session.c:1387 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:533 (sofia/external/123456789) State EXCHANGE_MEDIA going to sleep 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:467 (sofia/external/123456789) Running State Change CS_HANGUP 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:730 (sofia/external/123456789) Callstate Change EARLY -> HANGUP 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:732 (sofia/external/123456789) State HANGUP 2014-11-18 10:51:26.136684 [DEBUG] mod_sofia.c:413 Channel sofia/external/123456789 hanging up, cause: ORIGINATOR_CANCEL Anyone have any idea whats going on ? Thanks David From joelewhite at gmail.com Thu Nov 20 18:32:30 2014 From: joelewhite at gmail.com (Joel White) Date: Thu, 20 Nov 2014 10:32:30 -0500 Subject: [Freeswitch-users] Moving Voicemails In-Reply-To: <546BC48D.3020007@earthspike.net> References: <546A4215.2090509@earthspike.net> <0D35C356-F5B3-424A-B155-930959A9B210@jerris.com> <546BC48D.3020007@earthspike.net> Message-ID: I got it working good. The domain field as well as the file_path needed to be modified in the voicemail_msgs table. Thank you again :) On Tue, Nov 18, 2014 at 5:13 PM, John wrote: > I've just remembered why there were 2 updates of file_path. I moved the > voicemails from a box compiled from git into /usr/local/freeswitch to an > Ubuntu box with debian-style file paths. And I put a typo in the first > update, so had to do the IP-to-FQDN replace as a second transaction. > Anyway, glad you got it work despite my slightly misleading guidance! > > John > > > On 18/11/14 14:21, Joel White wrote: > > The change went smoothly. I really only used the update ***=replace > function to change out the IP to Domain > > Everything looks good > > On Tue, Nov 18, 2014 at 8:58 AM, Joel White wrote: > >> Thank you >> >> Changing the domain worked. >> >> Here is my file path as it has existed >> >> /usr/local/freeswitch/storage/voicemail/default/ >> 10.111.252.10/26343/msg_418e9e26-df74-11e3-b0cd-d9415cbdfb71.wav >> >> I just need the IP changed to the domain >> >> I will try the update of the file path today >> >> On Mon, Nov 17, 2014 at 1:56 PM, Michael Jerris wrote: >> >>> This does seem kind of clumsy... I'd be happy to accept a patch for >>> mod_voicemail change domain that moves files and updates db correctly if >>> anyone wants to give it a try. >>> >>> On Nov 17, 2014, at 1:44 PM, John wrote: >>> >>> I've done this and for exactly the same reason. I moved the files to the >>> domain-named path and updated the database using sqlite3. I found this in >>> my command history, but you should test first on a copy of the db. It's for >>> upgrading from 1.2.quite_old to 1.4.quite_recent. >>> >>> $ sqlite3 voicemail_default.db >>> sqlite> update voicemail_msgs set file_path='/var/lib' || >>> substr(replace(file_path,'172.16.12.34','pbx.somewhere.net'),11); >>> sqlite> update voicemail_msgs set >>> file_path=replace(file_path,'172.16.12.34','pbx.somewhere.net'); >>> sqlite> update voicemail_prefs set greeting_path='/var/lib' || >>> substr(replace(greeting_path,'172.16.12.34','pbx.somewhere.net'),11); >>> >>> I cannot remember what the '11' was for; I've changed the IP address and >>> FQDN so you might need to tune this value; check with SELECTs in place of >>> the UPDATEs. >>> >>> John >>> >>> On 17/11/14 18:08, Ken Rice wrote: >>> >>> You got it sqlite3 is the way to go... I doubt doing what you are trying >>> is heavily documented tho as its not something normally done. >>> >>> But all metadata for voicemail is in the sqlite db and that includes >>> file paths etc >>> >>> >>> On 11/17/14 10:07 AM, "Joel White" wrote: >>> >>> What is the easiest way to make this change? >>> >>> SQLite3 via command line? Or is there another way to manipulate the >>> database? >>> >>> On Mon, Nov 17, 2014 at 7:22 AM, Avi Marcus wrote: >>> >>> The path isn't dynamically generated based on the domain, it's stored >>> per-file in the database. >>> You can update the db or symlink the folders so they are available in >>> the old structure too. >>> -Avi >>> >>> On Mon, Nov 17, 2014 at 10:35 AM, Joel White >>> wrote: >>> >>> I initially configured my FreeSWITCH with an IP address, but now for >>> scalability and high availability I am changing to a domain name. I >>> successfully changed the system over, but was wondering how do I move the >>> voicemails to the new domain folder under voicemail? >>> >>> The obvious answer is to move the files, I already did that and >>> restarted the system. The existing voicemails and greetings are not >>> visible to FreeSWITCH when I call voicemail. >>> >>> How do I transfer them over? >>> >>> Thank you in advance for any light you may shed on this predicament >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/90871c66/attachment-0001.html From nasida at live.ru Thu Nov 20 21:31:06 2014 From: nasida at live.ru (Yuriy Nasida) Date: Thu, 20 Nov 2014 21:31:06 +0300 Subject: [Freeswitch-users] Trying to originate a call using ES. In-Reply-To: References: Message-ID: Hi, You can try to use loopback/[/context[/dialplan]] instead of user/1000 + originate_timeout It works for me. From: my.post at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Wed, 12 Nov 2014 23:26:20 +0600 Subject: [Freeswitch-users] Trying to originate a call using ES. Hello Everybody! I'm stuck trying to originate a call using event socket. Somewhere on the internet I've found some Perl script involving ESL. So I changed it a bit so I get an incoming call to outbound socket, look for destination number and trying to set up outgoing call using originate like so: $con->sendRecv("api originate user/1000 &park()") This works. The question is how can I control a unanswered timeout of this originate, say to playback some message to initial caller after 5 seconds ? I've tried setting the following variables: originate_timeout=5 leg_timeout=5 call_timeout=5 like so: $con->sendRecv("api originate {originate_timeout=5}user/1000 &park()") None of this working. The originated call keeps ringing. Please advise. Thanks. Regards, Pavel. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/1cf27b2d/attachment.html From kamil.nigmatullin at gmail.com Thu Nov 20 22:38:30 2014 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Fri, 21 Nov 2014 01:38:30 +0600 Subject: [Freeswitch-users] Save voicemail in opus format Message-ID: Hello. I want to save voicemail in opus codec so that client could listen it without transcoding. I set extension to opus instead of wav. Freeswitch started to write file with opus extension and it content of the file looks diffrent than wav. But I can playback this file even with unloaded mod_opus. VLC cannot play this file. How can this be explained? -- Kamil Nigmatullin Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/775b9faa/attachment.html From aqsyounas at gmail.com Thu Nov 20 23:17:51 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Fri, 21 Nov 2014 01:17:51 +0500 Subject: [Freeswitch-users] How i can disable tcp port in freeswitch.? Message-ID: Hi, Who can i disable tcp ports in freeswitch? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/6f739018/attachment.html From nneul at mst.edu Thu Nov 20 23:26:56 2014 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 20 Nov 2014 14:26:56 -0600 Subject: [Freeswitch-users] problem with eavesdrop() sending null RTP for 15-30 seconds Message-ID: <546E4E90.8070509@mst.edu> Seeing an interesting issue with sending an eavesdrop where the outbound call gets established, but no useful audio is transmitted until around 15-30 seconds into the call. Prior to that point, FS is transmitting all identical RTP frames with the payload containing: 24 1c 61 7a 7f 55 55 55 50 06 83 ff ff ff ff ff ff ff ff ff ff ff ff aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa af ff ..... ff This is minimal version of command that was being issued: originate sofia/internal/6679 at voice-dev.mst.edu &eavesdrop(ca887b3b-013f-4972-aa7c-37a7b132bf52) Reproduced with head of master as of a few minutes ago. Restarting FS and trying again, the FIRST attempted eavesdrop seems to not have the long delay, but the second one does. The 'dummy' payload appears to have the same content in subsequent requests. Interestingly, if I let the outbound call ring longer before I pick up, the delay appears to be reduced by the period of time I let the phone ring for. if I pick up immediately, 25-30 second audio delay approx, if I pick up after 25 seconds of ringing, I get only a couple second delay. Any idea where to start looking for where this buffering might be getting introduced? For the "only eavesdrop" case it's not as big of an issue, but when you switch over to barge, can't really have multi-second audio delay. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From jaugenstine at gmail.com Thu Nov 20 23:27:07 2014 From: jaugenstine at gmail.com (jonathan augenstine) Date: Thu, 20 Nov 2014 12:27:07 -0800 Subject: [Freeswitch-users] Save voicemail in opus format In-Reply-To: References: Message-ID: I will try to summarize to avoid previous conversations on this topic. Supporting the recording and playback of opus format files is difficult to impossible. This issue stems from the fact that opus is a stateful codec. This translates into the reality that attempting to simply inject the opus bitstream into a current call can result in a mismatch of the parameters giving an inaudible playback. The stateful codecs not only vary from call to call but can change within the same call. I have already gone down this road and it is a dead-end. Jonathan On Thu, Nov 20, 2014 at 11:38 AM, Kamil Nigmatullin < kamil.nigmatullin at gmail.com> wrote: > Hello. I want to save voicemail in opus codec so that client could listen > it without transcoding. > > I set extension to opus instead of wav. Freeswitch started to write file > with opus extension and it content of the file looks diffrent than wav. But > I can playback this file even with unloaded mod_opus. VLC cannot play this > file. > > How can this be explained? > > > -- > Kamil Nigmatullin > Skype: kamil.nigmatullin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/3334dc16/attachment.html From mike at jerris.com Thu Nov 20 23:38:50 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Nov 2014 15:38:50 -0500 Subject: [Freeswitch-users] How i can disable tcp port in freeswitch.? In-Reply-To: References: Message-ID: <3217598E-9D64-4F44-A4B9-EB043D152087@jerris.com> iptables is good at that, or other firewall software. Otherwise its a matter of correctly configuring the module that is listening on that port. What ports are you trying to "disable" > On Nov 20, 2014, at 3:17 PM, Aqs Younas wrote: > > Hi, > Who can i disable tcp ports in freeswitch? From vipkilla at gmail.com Thu Nov 20 23:41:20 2014 From: vipkilla at gmail.com (Vik Killa) Date: Thu, 20 Nov 2014 15:41:20 -0500 Subject: [Freeswitch-users] Presence question In-Reply-To: References: Message-ID: Do the lines support NOTIFY? On Wed, Nov 19, 2014 at 6:14 AM, Kamil Nigmatullin < kamil.nigmatullin at gmail.com> wrote: > One of our clients wants to switch on presence feature, but they have 140 > lines of FXS that does not support Publish/Subscribe function. Is there a > way to achive this using data in freeswitch (if Registered or Busy)? I know > that it is a silly question but they asked me to ask anyway, so will > anybody just confirm that it is not possible? > > -- > Kamil Nigmatullin > Skype: kamil.nigmatullin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/279cf578/attachment-0001.html From jalsot at gmail.com Thu Nov 20 23:50:32 2014 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Thu, 20 Nov 2014 21:50:32 +0100 Subject: [Freeswitch-users] Opus Codec Error In-Reply-To: <546DA4F4.3040108@kaymera.com> References: <546DA4F4.3040108@kaymera.com> Message-ID: Hello, Isn't it the same issue as reported in https://freeswitch.org/jira/browse/FS-6994 If yes, could you please give your comments in that ticket? Thanks, jalsot On Thu, Nov 20, 2014 at 9:23 AM, David Shemesh wrote: > Hope someone can help with this: > I am experiencing a strange issue using Opus. > The call is from a registered client to a landline through a sip Trunk. > Freeswitch is transcoding From Opus to PCMU and after about ~1min the > call drops with "switch_core_io.c:1282 Codec OPUS (STANDARD) decoder > error!" and the call gets disconnected. > It looks like this happens only when i enabling the jitterbuffer. > I was able to reproduce this using various Clients (Csipsimple, Jitsi) & > Sip trunks. it seems like its has something to do with freeswitch's > implementation of OPUS. > > FreeSWITCH Version 1.5.15b+git~20141112T004420Z~dd629c1516~64bit > > Dialplan: > > > > data="rtp_jitter_buffer_during_bridge=true" /> > > > > > data="effective_caller_id_name=${outbound_caller_id_name}" /> > data="effective_caller_id_number=${outbound_caller_id_number}" /> > > > > > > > > Freeswitch Log: > > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[opus:116:48000:20:0] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[PCMU:0:8000:20:64000] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3424 Audio Codec > Compare [PCMU:0:8000:20:64000] is saved as a near-match > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[GSM:3:8000:20:13200] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[PCMA:8:8000:20:64000] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[G722:9:8000:20:64000] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[AMR:96:8000:20:12200] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[iLBC:97:8000:30:13330] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[SPEEX:99:8000:20:24600] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[SPEEX:99:16000:20:42200] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec > Compare [PCMU:0:8000:10:64000]/[SPEEX:99:32000:20:44000] > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3315 Set > telephone-event payload to 101 > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3494 Substituting > codec PCMU at 10i@8000h > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:2343 Set Codec > sofia/external/123456789 PCMU/8000 10 ms 80 samples 64000 bits > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_codec.c:111 > sofia/external/123456789 Original read codec set to PCMU:0 > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3623 Set 2833 > dtmf send payload to 101 > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:4847 AUDIO RTP > [sofia/external/123456789] 10.0.50.160 port 23644 -> 212.179.176.54 port > 60036 codec: 0 ms: 10 > 2014-11-18 10:51:25.876680 [DEBUG] switch_rtp.c:3334 Starting timer > [soft] 80 bytes per 10ms > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5194 Set 2833 > dtmf send payload to 101 > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5200 Set 2833 > dtmf receive payload to 101 > 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5228 Set comfort > noise payload to 13 > 2014-11-18 10:51:25.876680 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/123456789! > 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3389 Send signal > sofia/internal/1000 at 192.168.10.150 [BREAK] > 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3393 > (sofia/external/123456789) Callstate Change RINGING -> EARLY > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_codec.c:246 > sofia/internal/1000 at 192.168.10.150 Restore previous codec opus:116. > 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_originate.c:3551 Originate > Resulted in Success: [sofia/external/123456789] > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal > sofia/external/123456789 [BREAK] > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal > sofia/internal/1000 at 192.168.10.150 [BREAK] > 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_bridge.c:1460 > (sofia/external/123456789) State Change CS_CONSUME_MEDIA -> > CS_EXCHANGE_MEDIA > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:1387 Send > signal sofia/external/123456789 [BREAK] > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:467 > (sofia/external/123456789) Running State Change CS_EXCHANGE_MEDIA > 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:533 > (sofia/external/123456789) State EXCHANGE_MEDIA > 2014-11-18 10:51:25.896682 [DEBUG] mod_sofia.c:592 SOFIA EXCHANGE_MEDIA > 2014-11-18 10:51:26.056685 [DEBUG] switch_rtp.c:5555 Correct ip/port > confirmed. > 2014-11-18 10:51:26.056685 [NOTICE] switch_core_io.c:1230 Activating > write resampler > 2014-11-18 10:51:26.056685 [DEBUG] switch_core_io.c:1458 Engaging Write > Buffer at 160 bytes to accommodate 320->160 > 2014-11-18 10:51:26.136684 [ERR] switch_core_io.c:1282 Codec OPUS > (STANDARD) decoder error! > 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:578 > sofia/external/123456789 ending bridge by request from write function > 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD > DONE [sofia/internal/1000 at 192.168.10.150] > 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal > sofia/external/123456789 [BREAK] > 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD > DONE [sofia/external/123456789] > 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal > sofia/internal/1000 at 192.168.10.150 [BREAK] > 2014-11-18 10:51:26.136684 [NOTICE] switch_ivr_bridge.c:751 Hangup > sofia/external/123456789 [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] > 2014-11-18 10:51:26.136684 [DEBUG] switch_channel.c:3215 Send signal > sofia/external/123456789 [KILL] > 2014-11-18 10:51:26.136684 [DEBUG] switch_core_session.c:1387 Send > signal sofia/external/123456789 [BREAK] > 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:533 > (sofia/external/123456789) State EXCHANGE_MEDIA going to sleep > 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:467 > (sofia/external/123456789) Running State Change CS_HANGUP > 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:730 > (sofia/external/123456789) Callstate Change EARLY -> HANGUP > 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:732 > (sofia/external/123456789) State HANGUP > 2014-11-18 10:51:26.136684 [DEBUG] mod_sofia.c:413 Channel > sofia/external/123456789 hanging up, cause: ORIGINATOR_CANCEL > > Anyone have any idea whats going on ? > > Thanks > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/d51d7cf4/attachment.html From danny.gershman at gmail.com Fri Nov 21 01:49:01 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Thu, 20 Nov 2014 22:49:01 +0000 Subject: [Freeswitch-users] HA recovery of custom channels variables Message-ID: I'm using some custom channel variables for storage and I noticed that when I do a recover they are not preserved because they are populated to the table 'recovery'. Is there something that I'm not doing to make this work? --Danny G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/217b2427/attachment.html From danny.gershman at gmail.com Fri Nov 21 01:50:53 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Thu, 20 Nov 2014 22:50:53 +0000 Subject: [Freeswitch-users] stereo mod_conference Message-ID: How do I enable stereo / 3d conferencing on mod_conference? This assumes I have a device that is capable of doing this. -- Danny G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/441b886f/attachment.html From danny.gershman at gmail.com Fri Nov 21 01:55:55 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Thu, 20 Nov 2014 22:55:55 +0000 Subject: [Freeswitch-users] mod_conference stereo? record stereo wav? (Stephen Dame) Message-ID: I'm interested in the same thing as well. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/a6c6fe86/attachment.html From danny.gershman at gmail.com Fri Nov 21 01:56:51 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Thu, 20 Nov 2014 22:56:51 +0000 (UTC) Subject: [Freeswitch-users] =?utf-8?q?mod=5Fconference_stereo=3F_record_st?= =?utf-8?q?ereo_wav=3F?= References: <04e901d00271$13bae5c0$3b30b140$@207me.com> Message-ID: I'm also interested. From brian at freeswitch.org Fri Nov 21 02:04:48 2014 From: brian at freeswitch.org (Brian West) Date: Thu, 20 Nov 2014 17:04:48 -0600 Subject: [Freeswitch-users] mod_conference stereo? record stereo wav? In-Reply-To: References: <04e901d00271$13bae5c0$3b30b140$@207me.com> Message-ID: File a bounty. ;) On Thu, Nov 20, 2014 at 4:56 PM, Danny Gershman wrote: > I'm also interested. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/3f9e1861/attachment-0001.html From mike at jerris.com Fri Nov 21 03:11:02 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Nov 2014 19:11:02 -0500 Subject: [Freeswitch-users] HA recovery of custom channels variables In-Reply-To: References: Message-ID: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> it depends when you are setting them. If they are not in the xml doc when you go to do recovery, they will not be re-populated. When/how are you setting them? > On Nov 20, 2014, at 5:49 PM, Danny Gershman wrote: > > I'm using some custom channel variables for storage and I noticed that when I do a recover they are not preserved because they are populated to the table 'recovery'. > > Is there something that I'm not doing to make this work? > > --Danny G > _________________________________________________________________________ From mike at jerris.com Fri Nov 21 03:13:25 2014 From: mike at jerris.com (Michael Jerris) Date: Thu, 20 Nov 2014 19:13:25 -0500 Subject: [Freeswitch-users] stereo mod_conference In-Reply-To: References: Message-ID: <08CBFD48-9744-4E91-921F-D0B97403E737@jerris.com> This requires a fairly recent version of openAL library (all the released ones are broken). We've been building our own packages using their git master for these as the ones in all the distros are too old. > On Nov 20, 2014, at 5:50 PM, Danny Gershman wrote: > > How do I enable stereo / 3d conferencing on mod_conference? This assumes I have a device that is capable of doing this. > > -- Danny G From olegstolyar at gmail.com Fri Nov 21 03:25:39 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Nov 2014 16:25:39 -0800 Subject: [Freeswitch-users] Conference recording Message-ID: Hi, I am planing to record conference using the auto-record param. My question is this: If the machine runs out of disk space, will FS just stop recording or will bad things like crash happen? Same question for logging. Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/65447863/attachment.html From msc at freeswitch.org Fri Nov 21 03:31:03 2014 From: msc at freeswitch.org (Michael Collins) Date: Thu, 20 Nov 2014 16:31:03 -0800 Subject: [Freeswitch-users] Conference recording In-Reply-To: References: Message-ID: Funny you should mention this. I was just today tinkering around with a system that has a relatively small disk (~36GB) and was moving some large files around. I accidentally filled up the disk. When I ran top I saw that FS was using ~80% CPU. I hopped on fs_cli and there was all sorts of red text flying across the screen. So, in my case FS didn't actually crash, although it certainly was not a happy camper. :) -MC On Thu, Nov 20, 2014 at 4:25 PM, Oleg Stolyar wrote: > Hi, > > I am planing to record conference using the auto-record param. > > My question is this: If the machine runs out of disk space, will FS just > stop recording or will bad things like crash happen? > > Same question for logging. > > Thank you > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/6a24bd1a/attachment.html From kheimerl at cs.berkeley.edu Fri Nov 21 04:07:59 2014 From: kheimerl at cs.berkeley.edu (Kurtis Heimerl) Date: Thu, 20 Nov 2014 17:07:59 -0800 Subject: [Freeswitch-users] Changing receiver IP on INVITE Message-ID: Hey FreeSWITCH Users, Interesting issue here. We're connecting to Nexmo, a major SIP provider. They do load balancing of their SIP servers; a set of four IPs behind sip.nexmo.com. I have a FS sip set up to register to their server. However, in INVITE, it seems to switch IPs. Basically it goes like this: US -> INVITE sip.nexmo.com (IP1) IP1 -> 407 AUTH US -> INVITE sip.nexmo.com (DIFFERENT IP2) nonce and auth stuff IP2 -> 904 FAIL Here's my config: ... How do we stop this from happening? Setting from-domain and proxy to an IP behind their load balancer fixes it, but that's clearly suboptimal. Why is FS resolving the sip.nexmo.com DNS twice? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/c3eeec66/attachment.html From olegstolyar at gmail.com Fri Nov 21 04:20:36 2014 From: olegstolyar at gmail.com (Oleg Stolyar) Date: Thu, 20 Nov 2014 17:20:36 -0800 Subject: [Freeswitch-users] Conference recording In-Reply-To: References: Message-ID: Thanks Michael! On Thu, Nov 20, 2014 at 4:31 PM, Michael Collins wrote: > Funny you should mention this. I was just today tinkering around with a > system that has a relatively small disk (~36GB) and was moving some large > files around. I accidentally filled up the disk. When I ran top I saw that > FS was using ~80% CPU. I hopped on fs_cli and there was all sorts of red > text flying across the screen. So, in my case FS didn't actually crash, > although it certainly was not a happy camper. :) > > -MC > > On Thu, Nov 20, 2014 at 4:25 PM, Oleg Stolyar > wrote: > >> Hi, >> >> I am planing to record conference using the auto-record param. >> >> My question is this: If the machine runs out of disk space, will FS just >> stop recording or will bad things like crash happen? >> >> Same question for logging. >> >> Thank you >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141120/4f54ff65/attachment.html From kamil.nigmatullin at gmail.com Fri Nov 21 08:25:41 2014 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Fri, 21 Nov 2014 11:25:41 +0600 Subject: [Freeswitch-users] Presence question In-Reply-To: References: Message-ID: Yes. I was actually quite surprised to see that FS works without PUBLISHes from clients' side. I can't help but think about it! That's really cool. 2014-11-21 2:41 GMT+06:00 Vik Killa : > Do the lines support NOTIFY? > > > On Wed, Nov 19, 2014 at 6:14 AM, Kamil Nigmatullin < > kamil.nigmatullin at gmail.com> wrote: > >> One of our clients wants to switch on presence feature, but they have 140 >> lines of FXS that does not support Publish/Subscribe function. Is there a >> way to achive this using data in freeswitch (if Registered or Busy)? I know >> that it is a silly question but they asked me to ask anyway, so will >> anybody just confirm that it is not possible? >> >> -- >> Kamil Nigmatullin >> Skype: kamil.nigmatullin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/c584371c/attachment-0001.html From danny.gershman at gmail.com Fri Nov 21 09:03:49 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 01:03:49 -0500 Subject: [Freeswitch-users] HA recovery of custom channels variables In-Reply-To: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> Message-ID: I have done a simple test, I call in with a softphone to a conference. I grab the uuid and I run uuid_setvar. On Thursday, November 20, 2014, Michael Jerris wrote: > it depends when you are setting them. If they are not in the xml doc when > you go to do recovery, they will not be re-populated. When/how are you > setting them? > > > On Nov 20, 2014, at 5:49 PM, Danny Gershman > wrote: > > > > I'm using some custom channel variables for storage and I noticed that > when I do a recover they are not preserved because they are populated to > the table 'recovery'. > > > > Is there something that I'm not doing to make this work? > > > > --Danny G > > _________________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/c1cbeca6/attachment.html From danny.gershman at gmail.com Fri Nov 21 09:08:03 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 01:08:03 -0500 Subject: [Freeswitch-users] stereo mod_conference In-Reply-To: <08CBFD48-9744-4E91-921F-D0B97403E737@jerris.com> References: <08CBFD48-9744-4E91-921F-D0B97403E737@jerris.com> Message-ID: Assuming I have this library. What do I need to do to configure this to make this work? Is there any documentation? How do the mod_verto demo work, is this using your custom build of OpenAL? On Thursday, November 20, 2014, Michael Jerris wrote: > This requires a fairly recent version of openAL library (all the released > ones are broken). We've been building our own packages using their git > master for these as the ones in all the distros are too old. > > > On Nov 20, 2014, at 5:50 PM, Danny Gershman > wrote: > > > > How do I enable stereo / 3d conferencing on mod_conference? This > assumes I have a device that is capable of doing this. > > > > -- Danny G > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/271feb3a/attachment.html From kamil.nigmatullin at gmail.com Fri Nov 21 10:57:35 2014 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Fri, 21 Nov 2014 13:57:35 +0600 Subject: [Freeswitch-users] Save voicemail in opus format In-Reply-To: References: Message-ID: I understand that. But opus is more like static codec in FS. If you will see source code of mod_opus there is an initial parameters hardcoded in module. and I bet it is not being changed during the call. Ok. What is the format that was recorded in file with opus extension? Why does it play with unloaded mod_opus? 2014-11-21 2:27 GMT+06:00 jonathan augenstine : > I will try to summarize to avoid previous conversations on this topic. > Supporting the recording and playback of opus format files is difficult to > impossible. This issue stems from the fact that opus is a stateful codec. > This translates into the reality that attempting to simply inject the opus > bitstream into a current call can result in a mismatch of the parameters > giving an inaudible playback. The stateful codecs not only vary from call > to call but can change within the same call. I have already gone down this > road and it is a dead-end. > > Jonathan > > On Thu, Nov 20, 2014 at 11:38 AM, Kamil Nigmatullin < > kamil.nigmatullin at gmail.com> wrote: > >> Hello. I want to save voicemail in opus codec so that client could listen >> it without transcoding. >> >> I set extension to opus instead of wav. Freeswitch started to write file >> with opus extension and it content of the file looks diffrent than wav. But >> I can playback this file even with unloaded mod_opus. VLC cannot play this >> file. >> >> How can this be explained? >> >> >> -- >> Kamil Nigmatullin >> Skype: kamil.nigmatullin >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/1e872b48/attachment.html From iskren.hadzhinedev at ikiji.com Fri Nov 21 14:49:24 2014 From: iskren.hadzhinedev at ikiji.com (Iskren Hadzhinedev) Date: Fri, 21 Nov 2014 13:49:24 +0200 Subject: [Freeswitch-users] DTMF detection doesn't work after a dialplan transfer Message-ID: <3472735.eGr0sI22Rx@axtroz-lenovo-ideapad-y510p> Hi everyone, using version 1.5.15b+git~20141109T085007Z~eb30491688~64bit (git eb30491 2014-11-09 08:50:07Z 64bit). When I use "transfer" in the dialplan to send someone to another context, DTMFs aren't detected. They pop up in the debug console, but voicemail seems to ignore them. Is there a way to workaround this? Thanks! Kind regards, -- Iskren Hadzhinedev From myforums.indra at gmail.com Fri Nov 21 10:24:04 2014 From: myforums.indra at gmail.com (indra sena) Date: Fri, 21 Nov 2014 12:54:04 +0530 Subject: [Freeswitch-users] Fwd: OverTime RAM usage keep inceasing In-Reply-To: References: Message-ID: Hi Team, I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. Do you have config changes to fix or to take as work around ? How can we reduce RAM utilization overtime ? Could you please provide some solution for this ? Thanks & Regards, Indra. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/0d393f11/attachment.html From k4kaleem at gmail.com Fri Nov 21 15:15:22 2014 From: k4kaleem at gmail.com (kaleem rehman) Date: Fri, 21 Nov 2014 12:15:22 +0000 Subject: [Freeswitch-users] how to play a file to recipient Message-ID: Hi All, I want to play a message to recipient to advise them what line call came in, in Avaya world this is known as Whisper / VDN of Origin Announcement. what I want to achieve is when call hits the Gateway its setup to route call to an extension, when extension answers the call they hear a wav file or a message via (Freeswitch SAY command) saying "customer service" or "Return" etc... thanks in advance, kaleem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/11731947/attachment-0001.html From ppyy at pubyun.com Fri Nov 21 17:59:58 2014 From: ppyy at pubyun.com (=?UTF-8?B?5b2t5YuH?=) Date: Fri, 21 Nov 2014 22:59:58 +0800 Subject: [Freeswitch-users] undefined symbol: luaL_prepbuffer in lua Message-ID: freeswitch at internal> 2014-11-21 22:57:35.379103 [ERR] mod_lua.cpp:203 error loading module 'socket.core' from file '/usr/lib/lua/5.1/socket/core.so': /usr/lib/lua/5.1/socket/core.so: undefined symbol: luaL_prepbuffer stack traceback: [C]: in ? [C]: in function 'require' /tmp/t.lua:1: in main chunk # cat /tmp/t.lua require "socket.core" local t =require('socket').tcp -- Peng Yong From krice at freeswitch.org Fri Nov 21 18:02:09 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 15:02:09 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <546f53f1cdc5_af4e7f3241001c8@ip-10-237-130-155.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/0641c451/attachment.html From mike at jerris.com Fri Nov 21 20:04:05 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Nov 2014 12:04:05 -0500 Subject: [Freeswitch-users] OverTime RAM usage keep inceasing In-Reply-To: References: Message-ID: The first thing i would do is confirm that this happens in 1.4.14 or master. If it does, please file a jira with as much detail as possible of what modules are loaded and what you are doing. You might want to try running under valgrind to see if there are large amounts of ram definitely lost in your scenario. > On Nov 21, 2014, at 2:24 AM, indra sena wrote: > > > Hi Team, > > I have observed that overtime RAM utilization/usage is keep increasing maybe due to memory leaks ? > > when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 hours load test and oberserved now it is using almost 11GB/12GB starts swap also increasing now. > > > Do you have config changes to fix or to take as work around ? > How can we reduce RAM utilization overtime ? > > > Could you please provide some solution for this ? > > Thanks & Regards, > Indra. > From mike at jerris.com Fri Nov 21 20:05:29 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Nov 2014 12:05:29 -0500 Subject: [Freeswitch-users] undefined symbol: luaL_prepbuffer in lua In-Reply-To: References: Message-ID: <9357E0F8-984E-4B85-AAF1-142163CC52B0@jerris.com> Looks like you are loading a lua module built against a different version of lua. > On Nov 21, 2014, at 9:59 AM, ?? wrote: > > freeswitch at internal> 2014-11-21 22:57:35.379103 [ERR] mod_lua.cpp:203 > error loading module 'socket.core' from file > '/usr/lib/lua/5.1/socket/core.so': > /usr/lib/lua/5.1/socket/core.so: undefined symbol: luaL_prepbuffer > stack traceback: > [C]: in ? > [C]: in function 'require' > /tmp/t.lua:1: in main chunk > > # cat /tmp/t.lua > require "socket.core" > local t =require('socket').tcp > > -- > Peng Yong From kamil.nigmatullin at gmail.com Fri Nov 21 20:12:19 2014 From: kamil.nigmatullin at gmail.com (Kamil Nigmatullin) Date: Fri, 21 Nov 2014 23:12:19 +0600 Subject: [Freeswitch-users] Redis database Message-ID: Dear all, Does mod_redis is used in FS for serving modules such as Limit? Can'u I get/set information to redis directly using API? Thanks -- Kamil Nigmatullin Skype: kamil.nigmatullin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/e522b622/attachment.html From krice at freeswitch.org Fri Nov 21 20:53:26 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 17:53:26 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) October 12th-18th Message-ID: <546f7c16891d1_26d379b3144169f@ip-10-156-165-173.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1z3PDLg FreeSWITCH Week in Review (Master Branch) October 12th-18th Hello, again. This week in the FreeSWITCH master branch we had 23 commits. Initial support for mod_verto on Windows was added this week as well as improvements to SIP ping generation and new commands in mod_snom. New features that were added: beb1d17 FS-6400 Improve sip ping generation by distributing them across an interval in mod_sofia [Jira: http://ift.tt/1F97ySH] f5f6d15 Add command ?action? with types ?reboot?, ?reset?, ?dialeddel?, ?misseddel?, and ?receiveddel? to mod_snom 15e9e68 FS-6927 #resolve #comment This display option ping times in the gateway status on sofia status gateways or individual gateway status output in mod_sofia [Jira: http://ift.tt/14XV8Cd] 7faf9f4 FS-6767 #comment Add initial support for mod_verto on Windows [Jira: http://ift.tt/1F97yCb] The following bugs were squashed: 2ca349a FS-6910 #resolve Multiple entry with the same first, last name or extension in the directory would only return 1 entry. Fix issue where group by would produce multiple row of count(*) result. Using distinct instead wouldn?t solve the issue in SQLITE because of a bug, so solution is to use a sub-select. [Jira: http://ift.tt/14XV8Ch] 412f214 Fix url encoding for snom remote commands (required to make # key work) 3cd0400 FS-6849 #resolve #comment scripts need to have ?from freeswitch import *? at the top in mod_python. I added it explicitly to compensate. [Jira: http://ift.tt/1F97yCd] 6b42e5a FS-6849 #comment change to ?import freeswitch? to only load it in mod_python. [Jira: http://ift.tt/1F97yCd] 2e10407 Actual fix for commit cff5209ca3582994dae1353372e2f91b345ab959 which was in the wrong place 1f9025d FS-6926 #resolve #comment Fix for Freeswitch not updating the jitterbuffer when it has to change the RTP packet size after a channel has already been set up. [Jira: http://ift.tt/14XV8Cj] In terms of stability these were the use cases that were fixed: 9bd3bd3 FS-6911 Segfault fix [Jira: http://ift.tt/1F97ySR] cff5209 Fix for a 3 to 7 year old leak of nua handle due to reference counting that effects all calls with auth/challenge on INVITE in mod_sofia Improvements in cross platform build supports: 2ca349a FS-6910 #resolve Multiple entry with the same first, last name or extension in the directory would only return 1 entry. Fix issue where group by would produce multiple row of count(*) result. Using distinct instead wouldn?t solve the issue in SQLITE because of a bug, so solution is to use a subselect. [Jira: http://ift.tt/14XV8Ch] 412f214 Fix url encoding for snom remote commands (required to make # key work) 3cd0400 FS-6849 #resolve #comment scripts need to have ?from freeswitch import *? at the top in mod_python. I added it explicitly to compensate. [Jira: http://ift.tt/1F97yCd] 6b42e5a FS-6849 #comment change to \?import freeswitch\? to only load it in mod_python. [Jira: http://ift.tt/1F97yCd] 2e10407 Actual fix for commit cff5209ca3582994dae1353372e2f91b345ab959 which was in the wrong place 1f9025d FS-6926 #resolve #comment Fix for Freeswitch not updating the jitterbuffer when it has to change the RTP packet size after a channel has already been set up. [Jira: http://ift.tt/14XV8Cj] ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/14558a8a/attachment.html From krice at freeswitch.org Fri Nov 21 21:13:04 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 18:13:04 +0000 Subject: [Freeswitch-users] FreeSWITCH Week in Review (Master Branch) October 19th-25th Message-ID: <546f80b07a744_8da3d2731456157@ip-10-183-3-7.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/14Y1L7B FreeSWITCH Week in Review (Master Branch) October 19th-25th Hello, again. This week in the FreeSWITCH master branch we had 8 commits. It was a very quiet week, but we did see the addition of the digits_dialed_filter and accompanying documentation. New features that were added: 8d720d5 FS-6940 #resolve #comment %FEATURE use the variable digits_dialed_filter to set regular expressions with () captures and anything matched will be replaced with X?s in the CDR [Jira: http://ift.tt/1xWtT4A] 711b0c7 FS-6940 docs and examples [Jira: http://ift.tt/1xWtT4A] 12b6940 Update jb command parser The following bugs were squashed: 3f0d6b3 FS-6756 lame_init_params must be called after setting all id3tag stuff, otherwise id3 tags will not be written. So, instead of calling it early, revert FS-3646 and add a check on free_context to really do lame stuff only if lame has been set ready, avoid seg faults in some corner cases. [Jira: http://ift.tt/1xWtT4C] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/ac6b9ad4/attachment-0001.html From danny.gershman at gmail.com Fri Nov 21 22:02:22 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 19:02:22 +0000 Subject: [Freeswitch-users] HA recovery of custom channels variables References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> Message-ID: I did another test with this, if I do a transfer right after I set the variable then it's available. Is there a way to propagate the channel variables into the xml doc without doing a transfer or executing inline? On Fri Nov 21 2014 at 1:03:49 AM Danny Gershman wrote: > I have done a simple test, I call in with a softphone to a conference. I > grab the uuid and I run uuid_setvar. > > On Thursday, November 20, 2014, Michael Jerris wrote: > >> it depends when you are setting them. If they are not in the xml doc >> when you go to do recovery, they will not be re-populated. When/how are >> you setting them? >> >> > On Nov 20, 2014, at 5:49 PM, Danny Gershman >> wrote: >> > >> > I'm using some custom channel variables for storage and I noticed that >> when I do a recover they are not preserved because they are populated to >> the table 'recovery'. >> > >> > Is there something that I'm not doing to make this work? >> > >> > --Danny G >> > >> _________________________________________________________________________ >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/88d2959e/attachment.html From igorolhovskiy at gmail.com Fri Nov 21 22:06:55 2014 From: igorolhovskiy at gmail.com (=?koi8-r?B?6cfP0tgg78zYyM/X08vJyg==?=) Date: Fri, 21 Nov 2014 21:06:55 +0200 Subject: [Freeswitch-users] HA recovery of custom channels variables In-Reply-To: References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> Message-ID: May be try to use external storage for channel vars, like redis? > 21 ????. 2014, ? 21:02, Danny Gershman ???????(?): > > I did another test with this, if I do a transfer right after I set the variable then it's available. Is there a way to propagate the channel variables into the xml doc without doing a transfer or executing inline? > >> On Fri Nov 21 2014 at 1:03:49 AM Danny Gershman wrote: >> I have done a simple test, I call in with a softphone to a conference. I grab the uuid and I run uuid_setvar. >> >>> On Thursday, November 20, 2014, Michael Jerris wrote: >>> it depends when you are setting them. If they are not in the xml doc when you go to do recovery, they will not be re-populated. When/how are you setting them? >>> >>> > On Nov 20, 2014, at 5:49 PM, Danny Gershman wrote: >>> > >>> > I'm using some custom channel variables for storage and I noticed that when I do a recover they are not preserved because they are populated to the table 'recovery'. >>> > >>> > Is there something that I'm not doing to make this work? >>> > >>> > --Danny G >>> > _________________________________________________________________________ >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/5b7e8e34/attachment.html From krice at freeswitch.org Fri Nov 21 22:26:39 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 19:26:39 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch) October 26th- November 2nd Message-ID: <546f91efa1e4a_aa0d91318702b9@ip-10-156-213-204.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1v3MNXh Freeswitch Week in Review (Master Branch) October 26th- November 2nd Hello, again. This week in the FreeSWITCH master branch we had 28 commits. Some of the features for this week include: rayo app now accepts optional comma separated list of JIDs or user names to steer incoming calls to specific rayo clients, new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. (It?s used in mod_conference to avoid reading audio while muted and possibly reduce some trans-coding load) , and the ability to log commands executed in mod_xml_rpc . New features that were added: a43e349 FS-6921 #resolve #comment rayo APP now accepts optional comma separated list of JIDs or user names to steer incoming calls to specific rayo clients Jira: http://ift.tt/1yzQC5L b25ae6a FS-5816 #resolve #comment Record-Completion-Cause added to session recording RECORD_STOP event and record_completion_cause channel variable added. Jira: http://ift.tt/1v3MPyt 0386db7 Add some asserts to catch buffer overflow d1e529a Add new hard_mute control to allow apps to request low level mute e.g. from the rtp stack level. Its used in mod_conference to avoid reading audio while muted and possibly reduce some transcoding load 49a3672 Add ability to log commands executed in mod_xml_rpc 3b9f0c3 FS-6927 #comment allow sub millisecond resolution for option ping times in mod_sofia Jira: http://ift.tt/14XV8Cd 8bfc4f2 DOCS-16 DOCS-15 and remember patches are welcome The following bugs were squashed: 57e8231 FS-6929 #resolve #comment Fix deadlock in mod_rayo Jira: http://ift.tt/1v3MOdQ 26af9c3 FS-6939 #resolve Fix for SIP ping generation missing sending options in mod_sofia Jira: http://ift.tt/1yzQEea bea7d8e FS-5853 #resolve #comment mod_rayo now reports record completion cause Jira: http://ift.tt/1v3MPyz f772b40 FS-6939 Fix for SIP ping generation occasionally missing sending options in mod_sofia Jira: http://ift.tt/1yzQEea 7581587 FS-6688 Fix for notify messages missing record-route tags after the second Subscribe message and also sometimes when sending a <state>early</state> in mod_sofia Jira: http://ift.tt/1yzQEuo 443ab8a FS-5949 Fix for error when resuming a call on hold (PJSIP and SILK) in mod_sofia Jira: http://ift.tt/1v3MOdW f85c824 FS-6949 Fix for OPUS codec channel being advertised as stereo in mod_opus Jira: http://ift.tt/1yzQEut ae0ab4c Overcome silly libedit bug when using screen or vt100 when using this .bashrc file 5488757 FS-6950 Fix for hard mute sticking in mod_conference Jira: http://ift.tt/1v3MPyD 43e6146 Translate dtmf directly to b leg in bypass call in mod_verto 52ae551 FS-6954 Fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia Jira: http://ift.tt/1yzQC5V 7ca4ac5 FS-5949 FS-6945 This patch improves the hold parsing and ignores connection address of 0 implying h Jira: http://ift.tt/1v3MOdW The complete list of commits can be found here:2014_10_26-2014_11_2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/b486d731/attachment.html From mike at jerris.com Fri Nov 21 22:31:51 2014 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Nov 2014 14:31:51 -0500 Subject: [Freeswitch-users] HA recovery of custom channels variables In-Reply-To: References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> Message-ID: <250EB7E4-80E3-4D4C-9A41-32CFE47592A6@jerris.com> The information is updated on bridge, unbridge, answer, and media negotiation. Basically any time the switch_core_recovery_track function is called. It would be fairly easy to add a way in dialplan for you to manually update, you could give that a try, and submit a patch. The other option wold be to do it on every dialplan execute, but I think that would be far to high a cost to do. The trick here is finding the right places to re-write the state information to the db, as there is cost involved, we don't want to do so in too many places. This is why your actual scenario matters. Mike > On Nov 21, 2014, at 2:02 PM, Danny Gershman wrote: > > I did another test with this, if I do a transfer right after I set the variable then it's available. Is there a way to propagate the channel variables into the xml doc without doing a transfer or executing inline? > > On Fri Nov 21 2014 at 1:03:49 AM Danny Gershman > wrote: > I have done a simple test, I call in with a softphone to a conference. I grab the uuid and I run uuid_setvar. > > On Thursday, November 20, 2014, Michael Jerris > wrote: > it depends when you are setting them. If they are not in the xml doc when you go to do recovery, they will not be re-populated. When/how are you setting them? > > > On Nov 20, 2014, at 5:49 PM, Danny Gershman > wrote: > > > > I'm using some custom channel variables for storage and I noticed that when I do a recover they are not preserved because they are populated to the table 'recovery'. > > > > Is there something that I'm not doing to make this work? > > > > --Danny G > > _________________________________________________________________________ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/b4fb42db/attachment-0001.html From danny.gershman at gmail.com Fri Nov 21 23:22:39 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 20:22:39 +0000 Subject: [Freeswitch-users] HA recovery of custom channels variables References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> <250EB7E4-80E3-4D4C-9A41-32CFE47592A6@jerris.com> Message-ID: Ok cool I think we are going to go the Redis route. We were using it to store information about the channel for our own application, essentially arbitrary information about our system. Regarding mod_redis... this doesn't already do this does it? What does mod_redis do? On Fri Nov 21 2014 at 2:32:38 PM Michael Jerris wrote: > The information is updated on bridge, unbridge, answer, and media > negotiation. Basically any time the switch_core_recovery_track function is > called. It would be fairly easy to add a way in dialplan for you to > manually update, you could give that a try, and submit a patch. The other > option wold be to do it on every dialplan execute, but I think that would > be far to high a cost to do. The trick here is finding the right places to > re-write the state information to the db, as there is cost involved, we > don't want to do so in too many places. This is why your actual scenario > matters. > > Mike > > On Nov 21, 2014, at 2:02 PM, Danny Gershman > wrote: > > I did another test with this, if I do a transfer right after I set the > variable then it's available. Is there a way to propagate the channel > variables into the xml doc without doing a transfer or executing inline? > > On Fri Nov 21 2014 at 1:03:49 AM Danny Gershman > wrote: > >> I have done a simple test, I call in with a softphone to a conference. I >> grab the uuid and I run uuid_setvar. >> >> On Thursday, November 20, 2014, Michael Jerris wrote: >> >>> it depends when you are setting them. If they are not in the xml doc >>> when you go to do recovery, they will not be re-populated. When/how are >>> you setting them? >>> >>> > On Nov 20, 2014, at 5:49 PM, Danny Gershman >>> wrote: >>> > >>> > I'm using some custom channel variables for storage and I noticed that >>> when I do a recover they are not preserved because they are populated to >>> the table 'recovery'. >>> > >>> > Is there something that I'm not doing to make this work? >>> > >>> > --Danny G >>> > >>> _________________________________________________________________________ >>> >>> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/a14c18e8/attachment.html From danny.gershman at gmail.com Sat Nov 22 00:17:09 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 21:17:09 +0000 Subject: [Freeswitch-users] conference play file position Message-ID: Is there a way to query the current file playback offset in a conference? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/9e19328b/attachment.html From krice at freeswitch.org Sat Nov 22 00:41:00 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 21:41:00 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch) November 2nd-8th Message-ID: <546fb16cc0a53_4d0033d3307382b@ip-10-69-135-107.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1Hwbe4X Freeswitch Week in Review (Master Branch) November 2nd-8th Hello, again. This week in the FreeSWITCH master branch we had 61 commits. Some of the many new features that were added this week are: the addition of console log level in mod_managed, support for per-module references directory in mod_managed , added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed , addition of unset app for the chatplan (and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms), and added a command to compile non-minified js file for testing in mod_verto. New features that were added: 7c0cf50 Set managedlist to return the value to the API stream and the log in mod_managed 4037e78 Support per-module references directory in mod_managed 10ebeba Add console log level in mod_managed 33cb950 Added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed 889b678 Added GetPtr to Util class for internal pointers extraction (very useful when using native api) in mod_managed 8478874 FS-6831 Addition of unset app for the chatplan and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms [Jira: http://ift.tt/1Hwa0Xw] bc767bb Adding rfc6598.auto and adding rfc6598 space to nat.auto acl. This is the NAT444 carrier grade nat space. 1190e59 FS-6965 Set the Accept-Language into a variable when received in mod_sofia [Jira: http://ift.tt/1uLG7M3] 94278b5 FS-5159 Allow enter and exit sounds to interrupt the MOH in a wait_mod conference 1944f9a FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min in mod_fifo [Jira: http://ift.tt/1Hwa0XA] 0f2816d Add command to compile non-minified js file for testing in mod_verto The following bugs were squashed: f87c335 Fix for issue with tracking agent availability 8f3c157 FS-6957 Fix for muting issues upon joining a conference in mod_conference [Jira: http://ift.tt/1uLG9Ue] 831832c FS-6890 Fix for FreeSWITCH forgets to send RE-INVITES [Jira: http://ift.tt/1uLG9Ug] 4eb5b38 Fix bug where re-invites needlessly re-init the codec and jb a497169 FS-6890 Additional work for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 9c1e603 FS-6954 #comment Fix for side effect of another fix for this bug. Please test this patch. [Jira: http://ift.tt/1yzQC5V] f66f2ca FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 415f82f FS-6954 Work toward fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia [Jira: http://ift.tt/1yzQC5V] 5ce5199 FS-6969 Fix for unwanted profile shutdown in mod_verto [Jira: http://ift.tt/1Hwa0XE] In terms of stability these were the use cases that were fixed: f3d089a Fix a crash when trying to remove a shadow directory in mod_managed Improvements in cross platform build supports: 5dee5ce FS-6953 Fix for build issues with OS X [Jira: http://ift.tt/1uLG9Um] a17be38 Add reconf target 300b8d8 FS-6973 Fix for changed Opal svn url breaking build of mod_opal [Jira: http://ift.tt/1Hwa1dY] The complete list of commits can be found here:2014_11_2-2014_11_9 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/5c1b9bf0/attachment.html From danny.gershman at gmail.com Sat Nov 22 00:58:33 2014 From: danny.gershman at gmail.com (Danny Gershman) Date: Fri, 21 Nov 2014 21:58:33 +0000 Subject: [Freeswitch-users] HA recovery of custom channels variables References: <9C3F5687-F533-41CF-A5DF-2BA0FE11DA9C@jerris.com> <250EB7E4-80E3-4D4C-9A41-32CFE47592A6@jerris.com> Message-ID: I also noticed a dialplan application that lets you do this. uuid_broadcast recovery_refresh:: What is this for? On Fri Nov 21 2014 at 3:22:39 PM Danny Gershman wrote: > Ok cool I think we are going to go the Redis route. We were using it to > store information about the channel for our own application, essentially > arbitrary information about our system. > > Regarding mod_redis... this doesn't already do this does it? What does > mod_redis do? > > On Fri Nov 21 2014 at 2:32:38 PM Michael Jerris wrote: > >> The information is updated on bridge, unbridge, answer, and media >> negotiation. Basically any time the switch_core_recovery_track function is >> called. It would be fairly easy to add a way in dialplan for you to >> manually update, you could give that a try, and submit a patch. The other >> option wold be to do it on every dialplan execute, but I think that would >> be far to high a cost to do. The trick here is finding the right places to >> re-write the state information to the db, as there is cost involved, we >> don't want to do so in too many places. This is why your actual scenario >> matters. >> >> Mike >> >> On Nov 21, 2014, at 2:02 PM, Danny Gershman >> wrote: >> >> I did another test with this, if I do a transfer right after I set the >> variable then it's available. Is there a way to propagate the channel >> variables into the xml doc without doing a transfer or executing inline? >> >> On Fri Nov 21 2014 at 1:03:49 AM Danny Gershman >> wrote: >> >>> I have done a simple test, I call in with a softphone to a conference. >>> I grab the uuid and I run uuid_setvar. >>> >>> On Thursday, November 20, 2014, Michael Jerris wrote: >>> >>>> it depends when you are setting them. If they are not in the xml doc >>>> when you go to do recovery, they will not be re-populated. When/how are >>>> you setting them? >>>> >>>> > On Nov 20, 2014, at 5:49 PM, Danny Gershman >>>> wrote: >>>> > >>>> > I'm using some custom channel variables for storage and I noticed >>>> that when I do a recover they are not preserved because they are populated >>>> to the table 'recovery'. >>>> > >>>> > Is there something that I'm not doing to make this work? >>>> > >>>> > --Danny G >>>> > ____________________________________________________________ >>>> _____________ >>>> >>>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/3c3a7bd5/attachment-0001.html From krice at freeswitch.org Sat Nov 22 01:54:31 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 21 Nov 2014 22:54:31 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch) November 9th-15th Message-ID: <546fc2a71c040_f1fcab932058147@ip-10-58-122-39.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/11M59RI Freeswitch Week in Review (Master Branch) November 9th-15th Hello, again. This week in the FreeSWITCH master branch we had 33 commits. Some of the new features for this week include: support for logging full timestamps with dialplan (defaults to old behavior unless requested), the addition of external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media, allow mod_http_cache to S3 services other than Amazon, and new configuration parameter add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). New features that were added: f175c71 FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested. Jira: http://ift.tt/11E7kpC fada4b8 FS-6977 Do not create freeswitch.serial if zrtp not enabled Jira: http://ift.tt/11E7kpG dd629c1 Add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media 0eefdca FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate ) Jira: http://ift.tt/11M58xc dd61232 FS-6979 Allow mod_http_cache to S3 services other than Amazon Jira: http://ift.tt/11E7i0R 826d428 FS-6992 Global configuration or maxplaybackrate and maxaveragebitrate from opus.conf.xml Jira: http://ift.tt/11M58xe e1c0ef5 New configuration parameter, add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). Improvements in cross platform build supports: 0f8b993 Fix mod_say_es_ar Makefile.am 07030c6 Fix compiler warning on unmatched return type The following bugs were squashed: 2f1b12f OPENZAP-232 Check for digits received on sangoma isdn stack to avoid delaying moving to the ring state if all digits are received at once in overlap dialing mode. 6b8d5b2 Fix release guard timer check in freetdm 34cf3b9 FS-6980 #resolve Fix for crash when using native recording on recordstop Jira: http://ift.tt/11E7i0U ab24bde FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device Jira: http://ift.tt/11M58xi a279bf3 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 0c68bb6 FS-6957 Fix for muting issues in mod_conference Jira: http://ift.tt/1uLG9Ue 33d37ce PLIV-13 Disable SSLv3 in libcurl for mod_httapi dbc5571 FS-6983 Wrap new curl TLS macro usage with ifdefs Jira: http://ift.tt/11E7i0W 0699ea8 Fixes for various issues 87a4670 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 82aa331 FS-6531: #resolve set to tag on auto answer notify Jira: http://ift.tt/11M58xk 75473a7 FS-6531: #resolve set to tag on uuid_phone_event notify Jira: http://ift.tt/11M58xk 07c5cc1 FS-7003 Fix infinite loop when output sent to server without SSML configured and repeat-times=0 Jira: http://ift.tt/11E7kG0 ? The complete list of commits can be found here:2014_11_9-2014_11_16 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/45705caa/attachment.html From steven.szeto at mitel.com Sat Nov 22 00:50:33 2014 From: steven.szeto at mitel.com (Szeto, Steven) Date: Fri, 21 Nov 2014 16:50:33 -0500 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: 1. Suppose you originate a call and park it: originate user/1001 &park 2. Now transfer the parked call: uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 inline When you do this, the parked caller hears nothing until the destination answers. How to you pass enough parameters to the uuid_transfer command to play ringback tone to the parked call during the transfer? -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141121/9821bbad/attachment.html From brian at freeswitch.org Sat Nov 22 18:13:11 2014 From: brian at freeswitch.org (Brian West) Date: Sat, 22 Nov 2014 09:13:11 -0600 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: transfer_ringback variable On Friday, November 21, 2014, Szeto, Steven wrote: > > > 1. Suppose you originate a call and park it: > > originate user/1001 &park > > > 2. Now transfer the parked call: > > uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 inline > > When you do this, the parked caller hears nothing until the destination > answers. > > How to you pass enough parameters to the uuid_transfer command to play > ringback tone to the parked call during the transfer? > > > > This e-mail (including any attachments) is for the sole use of the > intended recipient(s) and may contain information that is confidential > and/or protected by legal privilege. Any unauthorized review, use, copy, > disclosure or distribution of this e-mail is strictly prohibited. If you > are not the intended recipient, please notify Mitel immediately and destroy > all copies of this e-mail. Mitel does not accept any liability for breach > of security, error or virus that may result from the transmission of this > message. -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141122/881ef349/attachment.html From danb.lists at gmail.com Sat Nov 22 18:38:27 2014 From: danb.lists at gmail.com (Dan Christian Bogos) Date: Sat, 22 Nov 2014 16:38:27 +0100 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl Message-ID: <5470ADF3.5090502@gmail.com> Hey Guys, Was wondering if someone could help me with a working fifo.conf provisioned over xml_curl. I am trying to provision it but for some reason my configuration is not picked up, moreover if I am deleting autoload_configs/fifo.conf the mod_fifo will not even be loaded (although I am pushing it via xml_curl). Thanks in advance for any kind of tip! DanB From vetali100 at gmail.com Sat Nov 22 22:32:41 2014 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sat, 22 Nov 2014 11:32:41 -0800 Subject: [Freeswitch-users] Fwd: OverTime RAM usage keep inceasing In-Reply-To: References: Message-ID: Are you checking total RAM utilization or without cache/buffers? Can you run command "free" and paste the output? 2014-11-20 23:24 GMT-08:00 indra sena : > > Hi Team, > > I have observed that overtime RAM utilization/usage is keep increasing > maybe due to memory leaks ? > > when I put load initially it starts with 5GB/12GM RAM, I have kept for 15 > hours load test and oberserved now it is using almost 11GB/12GB starts swap > also increasing now. > > > Do you have config changes to fix or to take as work around ? > How can we reduce RAM utilization overtime ? > > > Could you please provide some solution for this ? > > Thanks & Regards, > Indra. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141122/32cec704/attachment-0001.html From amkusmirek at gmail.com Sun Nov 23 01:42:03 2014 From: amkusmirek at gmail.com (=?UTF-8?Q?Adam_Ku=C5=9Bmirek?=) Date: Sat, 22 Nov 2014 23:42:03 +0100 Subject: [Freeswitch-users] uuid_send_dtmf problem with rfc2833 Message-ID: Hello, I have a problem with sending dtmf events (rfc2833) with uuid_send_dtmf api command. FreeSWITCH Version 1.5.15b+git~20141118T231404Z~df423b88d6~64bit (git df423b8 2014-11-18 23:14:04Z 64bit) First i originate the call and park it with the command: originate {origination_caller_id_name=test}sofia/gateway/test/ xxxxxxxx at yyy.zzz.com &park() The other side answers the call. I can see that rfc 2833 is negotiated in SDPs. SIP/2.0 200 Ok Via: SIP/2.0/UDP 46.174.232.238:5070;rport;branch=z9hG4bKFmQvapcNNp44m From: "..." ;tag=35FN6HXZ3ry2F To: ;tag=f0313ac544a16a82aee0b Contact: sip:... at ...:5060 Call-ID: f7b0d42d-ed32-1232-06a5-c03fd5651f3e CSeq: 68010140 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 211 v=0 o=xxx 1416695524 1416695524 IN IP4 yyyy s=SIP Call c=IN IP4 yyyy t=0 0 m=audio 27792 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 Then I try to send DTMF freeswitch at internal> uuid_send_dtmf 6022b884-7297-11e4-8682-fd8246740d0e 12345 -ERR no reply 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254 at sip.12voip.com send dtmf digit=1 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254 at sip.12voip.com send dtmf digit=2 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254 at sip.12voip.com send dtmf digit=3 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254 at sip.12voip.com send dtmf digit=4 ms=250 samples=2000 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ 12122777254 at sip.12voip.com send dtmf digit=5 ms=250 samples=2000 Unfortunatelly it doesn't work and I can't see any rtpevents in wireshark. If I change dtmf-type in profile to info, i can see sip info packets. Please help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141122/075108ef/attachment.html From idokan at gmail.com Sun Nov 23 13:12:16 2014 From: idokan at gmail.com (ik) Date: Sun, 23 Nov 2014 12:12:16 +0200 Subject: [Freeswitch-users] Changing receiver IP on INVITE In-Reply-To: References: Message-ID: AFAIK, It's part of the DNS load balancing "thingy". It's a feature not a bug. You place a weight for connections per specific DNS. Here are the records for nexmo: $ dig srv _sip._udp.sip.nexmo.com +short 10 10 5060 sip1.nexmo.com. 20 10 5060 sip3.nexmo.com. 10 10 5060 sip2.nexmo.com. 20 10 5060 sip4.nexmo.com. It's up to nexmo to handle this properly, because it's their settings. On Fri, Nov 21, 2014 at 3:07 AM, Kurtis Heimerl wrote: > Hey FreeSWITCH Users, > > Interesting issue here. We're connecting to Nexmo, a major SIP provider. > They do load balancing of their SIP servers; a set of four IPs behind > sip.nexmo.com. > > I have a FS sip set up to register to their server. However, in INVITE, it > seems to switch IPs. Basically it goes like this: > > US -> INVITE sip.nexmo.com (IP1) > IP1 -> 407 AUTH > US -> INVITE sip.nexmo.com (DIFFERENT IP2) nonce and auth stuff > IP2 -> 904 FAIL > > Here's my config: > > > > > > > > > > > > ... > > How do we stop this from happening? Setting from-domain and proxy to an IP > behind their load balancer fixes it, but that's clearly suboptimal. Why is > FS resolving the sip.nexmo.com DNS twice? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141123/e59fa4ca/attachment.html From karl-theo_hofer at inteli-sim.com Sun Nov 23 18:48:35 2014 From: karl-theo_hofer at inteli-sim.com (kthofer) Date: Sun, 23 Nov 2014 16:48:35 +0100 Subject: [Freeswitch-users] Strange behaviour when calling 2 destinations in parallel Message-ID: <547201D3.1070005@inteli-sim.com> Hi There one incomming call shall originate 2 outgoing calls in parallel. one outgoing channel shall be terminated to a SIP/PSTN gateway the other one shall be terminated to a sip extension connected to a OPensips server. Both extensions (sip and PSTN) are ringing but when one of them answers the call, I get the loose_race from the not answered channel and then all calls are being released and everything collapse. any idea what this can be? I uploaded the whole call on to past bin please can someone have a look what the reason can be that the calls are releasing. past bin link http://pastebin.freeswitch.org/23613 -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer at inteli-sim.com From youssef.elouam at outlook.com Sun Nov 23 19:58:18 2014 From: youssef.elouam at outlook.com (youssef elouam) Date: Sun, 23 Nov 2014 16:58:18 +0000 Subject: [Freeswitch-users] mod_sofia.c:2219 CODEC NEGOTIATION ERROR Message-ID: Dear All ; many thanks in advance for your help. I have problem of CODEC NEGOTIATION in my freeswitch installed on Raspberry PI board. following is logs I have : 2014-11-23 16:52:16.832177 [ERR] mod_sofia.c:2219 CODEC NEGOTIATION ERROR. SDP:v=0o=- 3625750336 3625750336 IN IP4 192.168.1.5s=pjmediac=IN IP4 192.168.1.5t=0 0m=audio 4006 RTP/AVP 99 0 8 101c=IN IP4 192.168.1.5a=rtpmap:99 SILK/24000a=fmtp:99 useinbandfec=0a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtcp:4007 IN IP4 192.168.1.5 freeswitch at internal> versionFreeSWITCH Version 1.5.15b+git~20141024T202644Z~12b6940644~32bit (git 12b6940 2014-10-24 20:26:44Z 32bit) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141123/e7e8cd27/attachment.html From krice at freeswitch.org Sun Nov 23 22:14:57 2014 From: krice at freeswitch.org (Ken Rice) Date: Sun, 23 Nov 2014 13:14:57 -0600 Subject: [Freeswitch-users] mod_sofia.c:2219 CODEC NEGOTIATION ERROR In-Reply-To: Message-ID: You need to be a bit more specific then that.. That?s only showing the SDP... But on the Raspberry PI there is a known GCC bug if you are using an older version of GCC see http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105974.html On 11/23/14 10:58 AM, "youssef elouam" wrote: > Dear All ; > > many thanks in advance for your help. I have problem of CODEC NEGOTIATION in > my freeswitch installed on Raspberry PI board. > > following is logs I have : > > 2014-11-23 16:52:16.832177 [ERR] mod_sofia.c:2219 CODEC NEGOTIATION ERROR. > SDP: > v=0 > o=- 3625750336 3625750336 IN IP4 192.168.1.5 > s=pjmedia > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 4006 RTP/AVP 99 0 8 101 > c=IN IP4 192.168.1.5 > a=rtpmap:99 SILK/24000 > a=fmtp:99 useinbandfec=0 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtcp:4007 IN IP4 192.168.1.5 > > > freeswitch at internal> version > FreeSWITCH Version 1.5.15b+git~20141024T202644Z~12b6940644~32bit (git 12b6940 > 2014-10-24 20:26:44Z 32bit) > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141123/1125299e/attachment-0001.html From ssinyagin at gmail.com Mon Nov 24 00:31:30 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sun, 23 Nov 2014 22:31:30 +0100 Subject: [Freeswitch-users] Strange behaviour when calling 2 destinations in parallel In-Reply-To: <547201D3.1070005@inteli-sim.com> References: <547201D3.1070005@inteli-sim.com> Message-ID: Probably you forgot to set ignore early media to true. On Nov 23, 2014 4:49 PM, "kthofer" wrote: > Hi There > one incomming call shall originate 2 outgoing calls in parallel. > one outgoing channel shall be terminated to a SIP/PSTN gateway > the other one shall be terminated to a sip extension connected to a > OPensips server. > > Both extensions (sip and PSTN) are ringing but when one of them answers > the call, I get the loose_race from the not answered channel and then > all calls are being released > and everything collapse. > any idea what this can be? > > I uploaded the whole call on to past bin please can someone have a look > what the reason can be that the calls are releasing. > past bin link > http://pastebin.freeswitch.org/23613 > > > -- > With best regards > > Karl Theo Hofer > > M: +46 7030 22178 > E: karl-theo_hofer at inteli-sim.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141123/1cfe681c/attachment.html From steven.szeto at mitel.com Mon Nov 24 04:47:26 2014 From: steven.szeto at mitel.com (Szeto, Steven) Date: Sun, 23 Nov 2014 20:47:26 -0500 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: Thanks, Brian. Can anyone give an example of the syntax to set the transfer_ringback variable within the uuid_transfer command? I've tried a few combos that either result in a syntax error or the call dropping. I think it is supposed to look something like this: uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff [ transfer_ringback=$${us-ring}]bridge:user/1002 inline On Sat, Nov 22, 2014 at 10:13 AM, Brian West wrote: > transfer_ringback variable > > > On Friday, November 21, 2014, Szeto, Steven > wrote: > >> >> >> 1. Suppose you originate a call and park it: >> >> originate user/1001 &park >> >> >> 2. Now transfer the parked call: >> >> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 inline >> >> When you do this, the parked caller hears nothing until the destination >> answers. >> >> How to you pass enough parameters to the uuid_transfer command to play >> ringback tone to the parked call during the transfer? >> >> >> >> This e-mail (including any attachments) is for the sole use of the >> intended recipient(s) and may contain information that is confidential >> and/or protected by legal privilege. Any unauthorized review, use, copy, >> disclosure or distribution of this e-mail is strictly prohibited. If you >> are not the intended recipient, please notify Mitel immediately and destroy >> all copies of this e-mail. Mitel does not accept any liability for breach >> of security, error or virus that may result from the transmission of this >> message. > > > > -- > > *Brian West* > brian at freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141123/312d803f/attachment.html From ssinyagin at gmail.com Mon Nov 24 12:05:56 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 24 Nov 2014 10:05:56 +0100 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: you just execute uuid_setvar [value] before executing uuid_transfer On Mon, Nov 24, 2014 at 2:47 AM, Szeto, Steven wrote: > Thanks, Brian. > > Can anyone give an example of the syntax to set the transfer_ringback > variable within the uuid_transfer command? > > I've tried a few combos that either result in a syntax error or the call > dropping. > > I think it is supposed to look something like this: > > uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff [ > transfer_ringback=$${us-ring}]bridge:user/1002 inline > > > On Sat, Nov 22, 2014 at 10:13 AM, Brian West wrote: > >> transfer_ringback variable >> >> >> On Friday, November 21, 2014, Szeto, Steven >> wrote: >> >>> >>> >>> 1. Suppose you originate a call and park it: >>> >>> originate user/1001 &park >>> >>> >>> 2. Now transfer the parked call: >>> >>> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 >>> inline >>> >>> When you do this, the parked caller hears nothing until the destination >>> answers. >>> >>> How to you pass enough parameters to the uuid_transfer command to play >>> ringback tone to the parked call during the transfer? >>> >>> >>> >>> This e-mail (including any attachments) is for the sole use of the >>> intended recipient(s) and may contain information that is confidential >>> and/or protected by legal privilege. Any unauthorized review, use, copy, >>> disclosure or distribution of this e-mail is strictly prohibited. If you >>> are not the intended recipient, please notify Mitel immediately and destroy >>> all copies of this e-mail. Mitel does not accept any liability for breach >>> of security, error or virus that may result from the transmission of this >>> message. >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> >> *Twitter: @FreeSWITCH , @briankwest* >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > This e-mail (including any attachments) is for the sole use of the > intended recipient(s) and may contain information that is confidential > and/or protected by legal privilege. Any unauthorized review, use, copy, > disclosure or distribution of this e-mail is strictly prohibited. If you > are not the intended recipient, please notify Mitel immediately and destroy > all copies of this e-mail. Mitel does not accept any liability for breach > of security, error or virus that may result from the transmission of this > message. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/8f9ed147/attachment-0001.html From italorossib at gmail.com Mon Nov 24 14:56:16 2014 From: italorossib at gmail.com (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 24 Nov 2014 08:56:16 -0300 Subject: [Freeswitch-users] uuid_send_dtmf problem with rfc2833 In-Reply-To: References: Message-ID: I also had this problem, a workaround is send any audio before sending the dtmf, it'll work. Does anyone know why this happen? On Sat, Nov 22, 2014 at 7:42 PM, Adam Ku?mirek wrote: > Hello, > > I have a problem with sending dtmf events (rfc2833) with uuid_send_dtmf > api command. > > FreeSWITCH Version 1.5.15b+git~20141118T231404Z~df423b88d6~64bit (git > df423b8 2014-11-18 23:14:04Z 64bit) > > First i originate the call and park it with the command: > > originate {origination_caller_id_name=test}sofia/gateway/test/ > xxxxxxxx at yyy.zzz.com &park() > > The other side answers the call. I can see that rfc 2833 is negotiated in > SDPs. > > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 46.174.232.238:5070;rport;branch=z9hG4bKFmQvapcNNp44m > From: "..." ;tag=35FN6HXZ3ry2F > To: ;tag=f0313ac544a16a82aee0b > Contact: sip:... at ...:5060 > Call-ID: f7b0d42d-ed32-1232-06a5-c03fd5651f3e > CSeq: 68010140 INVITE > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Type: application/sdp > Content-Length: 211 > > v=0 > o=xxx 1416695524 1416695524 IN IP4 yyyy > s=SIP Call > c=IN IP4 yyyy > t=0 0 > m=audio 27792 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > > Then I try to send DTMF > > freeswitch at internal> uuid_send_dtmf 6022b884-7297-11e4-8682-fd8246740d0e > 12345 > -ERR no reply > > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ > 12122777254 at sip.12voip.com send dtmf > digit=1 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ > 12122777254 at sip.12voip.com send dtmf > digit=2 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ > 12122777254 at sip.12voip.com send dtmf > digit=3 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ > 12122777254 at sip.12voip.com send dtmf > digit=4 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/ > 12122777254 at sip.12voip.com send dtmf > digit=5 ms=250 samples=2000 > > > Unfortunatelly it doesn't work and I can't see any rtpevents in wireshark. > > If I change dtmf-type in profile to info, i can see sip info packets. > > > Please help > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ?talo Rossi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/a1b60103/attachment.html From vipkilla at gmail.com Mon Nov 24 15:56:59 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 24 Nov 2014 07:56:59 -0500 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: <5470ADF3.5090502@gmail.com> References: <5470ADF3.5090502@gmail.com> Message-ID: Make sure you have XML_CURL configured as follows; FS will send something like this to your script: http://localhost/index.php?FreeSWITCH-Hostname=fs01&FreeSWITCH-Switchname=fs-01&hostname=fs-01§ion=configuration&key_name=name&key_value=fifo.conf Write a script to parse the headers and generate the appropriate XML config. On Sat, Nov 22, 2014 at 10:38 AM, Dan Christian Bogos wrote: > Hey Guys, > > Was wondering if someone could help me with a working fifo.conf > provisioned over xml_curl. > > I am trying to provision it but for some reason my configuration is not > picked up, moreover if I am deleting autoload_configs/fifo.conf the > mod_fifo will not even be loaded (although I am pushing it via xml_curl). > > Thanks in advance for any kind of tip! > > DanB > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/cf00f990/attachment.html From steveayre at gmail.com Mon Nov 24 16:58:43 2014 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 24 Nov 2014 13:58:43 +0000 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: References: <5470ADF3.5090502@gmail.com> Message-ID: Also make sure you're wrapping the config you send back in the correct tags. On 24 November 2014 at 12:56, Vik Killa wrote: > Make sure you have XML_CURL configured as follows; > > > > > bindings="configuration" /> > > > > > FS will send something like this to your script: > > > http://localhost/index.php?FreeSWITCH-Hostname=fs01&FreeSWITCH-Switchname=fs-01&hostname=fs-01§ion=configuration&key_name=name&key_value=fifo.conf > > Write a script to parse the headers and generate the appropriate XML > config. > > > > > > > On Sat, Nov 22, 2014 at 10:38 AM, Dan Christian Bogos < > danb.lists at gmail.com> wrote: > >> Hey Guys, >> >> Was wondering if someone could help me with a working fifo.conf >> provisioned over xml_curl. >> >> I am trying to provision it but for some reason my configuration is not >> picked up, moreover if I am deleting autoload_configs/fifo.conf the >> mod_fifo will not even be loaded (although I am pushing it via xml_curl). >> >> Thanks in advance for any kind of tip! >> >> DanB >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/9c8301e6/attachment.html From steven.szeto at mitel.com Mon Nov 24 17:01:07 2014 From: steven.szeto at mitel.com (Szeto, Steven) Date: Mon, 24 Nov 2014 09:01:07 -0500 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: Thanks Stanislav & Brian. It worked. Here's a summary of the call sequence: 1. Originate a call and park it: originate user/1001 &park 2. Set the transfer_ringback variable to US ringback: uuid_setvar e121f9cb-f9a2-43fa-a61e-2a9a7cd5df00 transfer_ringback %(2000,4000,440.0,480.0) 3. Now transfer the parked call: uuid_transfer e121f9cb-f9a2-43fa-a61e-2a9a7cd5df00 bridge:user/1002 inline On Mon, Nov 24, 2014 at 4:05 AM, Stanislav Sinyagin wrote: > you just execute > uuid_setvar [value] > before executing uuid_transfer > > On Mon, Nov 24, 2014 at 2:47 AM, Szeto, Steven > wrote: > >> Thanks, Brian. >> >> Can anyone give an example of the syntax to set the transfer_ringback >> variable within the uuid_transfer command? >> >> I've tried a few combos that either result in a syntax error or the call >> dropping. >> >> I think it is supposed to look something like this: >> >> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff [ >> transfer_ringback=$${us-ring}]bridge:user/1002 inline >> >> >> On Sat, Nov 22, 2014 at 10:13 AM, Brian West >> wrote: >> >>> transfer_ringback variable >>> >>> >>> On Friday, November 21, 2014, Szeto, Steven >>> wrote: >>> >>>> >>>> >>>> 1. Suppose you originate a call and park it: >>>> >>>> originate user/1001 &park >>>> >>>> >>>> 2. Now transfer the parked call: >>>> >>>> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 >>>> inline >>>> >>>> When you do this, the parked caller hears nothing until the destination >>>> answers. >>>> >>>> How to you pass enough parameters to the uuid_transfer command to play >>>> ringback tone to the parked call during the transfer? >>>> >>>> >>>> >>>> This e-mail (including any attachments) is for the sole use of the >>>> intended recipient(s) and may contain information that is confidential >>>> and/or protected by legal privilege. Any unauthorized review, use, copy, >>>> disclosure or distribution of this e-mail is strictly prohibited. If you >>>> are not the intended recipient, please notify Mitel immediately and destroy >>>> all copies of this e-mail. Mitel does not accept any liability for breach >>>> of security, error or virus that may result from the transmission of this >>>> message. >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> This e-mail (including any attachments) is for the sole use of the >> intended recipient(s) and may contain information that is confidential >> and/or protected by legal privilege. Any unauthorized review, use, copy, >> disclosure or distribution of this e-mail is strictly prohibited. If you >> are not the intended recipient, please notify Mitel immediately and destroy >> all copies of this e-mail. Mitel does not accept any liability for breach >> of security, error or virus that may result from the transmission of this >> message. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/a9652c2c/attachment-0001.html From wstephen80 at gmail.com Mon Nov 24 17:06:03 2014 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 24 Nov 2014 15:06:03 +0100 Subject: [Freeswitch-users] RTP Packet Loss Troubleshooting In-Reply-To: References: Message-ID: Do you see all cores (with 'htop') at 40% of usage or there is one near 100%? On Thu, Oct 16, 2014 at 2:39 PM, Linux Vince wrote: > I have installed freeswitch and testing with SiPP > > FreeSwitch was easily able to handle 1700 concurrent calls. > > However when it starts going near 2000 my RTP packets are being lost and > audio is totally gone in some cases. > > How do i troubleshoot this ? And how to tune RTP performance for > FreeSwitch. > > I checked that CPU is not fully utilized and is somewhere around 40% range > and memory is only at 10%. > > Also network interface is not fully utilized as well as i was able to > transfer files from server and interface was able to handle much more > bandwidth than it was running. > > Please help. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/c5df425c/attachment.html From mike at jerris.com Mon Nov 24 17:55:24 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 09:55:24 -0500 Subject: [Freeswitch-users] uuid_send_dtmf problem with rfc2833 In-Reply-To: References: Message-ID: <7C6DD450-F7A3-4594-B25E-09835D1322AD@jerris.com> maybe nat issue? It's impossible to guess with this little information. > On Nov 24, 2014, at 6:56 AM, ?talo Rossi wrote: > > I also had this problem, a workaround is send any audio before sending the dtmf, it'll work. > > Does anyone know why this happen? > > On Sat, Nov 22, 2014 at 7:42 PM, Adam Ku?mirek > wrote: > Hello, > > I have a problem with sending dtmf events (rfc2833) with uuid_send_dtmf api command. > > FreeSWITCH Version 1.5.15b+git~20141118T231404Z~df423b88d6~64bit (git df423b8 2014-11-18 23:14:04Z 64bit) > > First i originate the call and park it with the command: > > originate {origination_caller_id_name=test}sofia/gateway/test/xxxxxxxx at yyy.zzz.com &park() > > The other side answers the call. I can see that rfc 2833 is negotiated in SDPs. > > SIP/2.0 200 Ok > Via: SIP/2.0/UDP 46.174.232.238:5070;rport;branch=z9hG4bKFmQvapcNNp44m > From: "..." ;tag=35FN6HXZ3ry2F > To: >;tag=f0313ac544a16a82aee0b > Contact: sip:... at ...:5060 > Call-ID: f7b0d42d-ed32-1232-06a5-c03fd5651f3e > CSeq: 68010140 INVITE > Server: (Very nice Sip Registrar/Proxy Server) > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE > Content-Type: application/sdp > Content-Length: 211 > > v=0 > o=xxx 1416695524 1416695524 IN IP4 yyyy > s=SIP Call > c=IN IP4 yyyy > t=0 0 > m=audio 27792 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > > > Then I try to send DTMF > > freeswitch at internal> uuid_send_dtmf 6022b884-7297-11e4-8682-fd8246740d0e 12345 > -ERR no reply > > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com send dtmf > digit=1 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com send dtmf > digit=2 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com send dtmf > digit=3 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com send dtmf > digit=4 ms=250 samples=2000 > 2014-11-22 23:32:21.888589 [DEBUG] switch_core_io.c:1976 sofia/external/12122777254 at sip.12voip.com send dtmf > digit=5 ms=250 samples=2000 > > > Unfortunatelly it doesn't work and I can't see any rtpevents in wireshark. > > If I change dtmf-type in profile to info, i can see sip info packets. > > > Please help > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/fa4c6322/attachment.html From mike at jerris.com Mon Nov 24 17:56:50 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 09:56:50 -0500 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: References: <5470ADF3.5090502@gmail.com> Message-ID: Make sure you are loading mod_xml_curl before any modules that need to use it to pull config. > On Nov 24, 2014, at 8:58 AM, Steven Ayre wrote: > > Also make sure you're wrapping the config you send back in the correct tags. > > On 24 November 2014 at 12:56, Vik Killa > wrote: > Make sure you have XML_CURL configured as follows; > > > > > > > > > > FS will send something like this to your script: > > http://localhost/index.php?FreeSWITCH-Hostname=fs01&FreeSWITCH-Switchname=fs-01&hostname=fs-01§ion=configuration&key_name=name&key_value=fifo.conf > > Write a script to parse the headers and generate the appropriate XML config. > > On Sat, Nov 22, 2014 at 10:38 AM, Dan Christian Bogos > wrote: > Hey Guys, > > Was wondering if someone could help me with a working fifo.conf > provisioned over xml_curl. > > I am trying to provision it but for some reason my configuration is not > picked up, moreover if I am deleting autoload_configs/fifo.conf the > mod_fifo will not even be loaded (although I am pushing it via xml_curl). > > Thanks in advance for any kind of tip! > > DanB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/3f509b0a/attachment.html From mike at jerris.com Mon Nov 24 18:00:27 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 10:00:27 -0500 Subject: [Freeswitch-users] RTP Packet Loss Troubleshooting In-Reply-To: References: Message-ID: <184C1B38-47AE-4BB2-89F2-100C5EB71330@jerris.com> Are you sure you are not on a 100mb or 1/2 duplex link in your network path? These sorts of issues typically get down to network hardware, network drivers, os cache settings, and that sort of thing. Lots of devices and hardware do not care for high packets per second of voip. This could just be a junk switch in the mix. > On Oct 16, 2014, at 8:39 AM, Linux Vince wrote: > > I have installed freeswitch and testing with SiPP > > FreeSwitch was easily able to handle 1700 concurrent calls. > > However when it starts going near 2000 my RTP packets are being lost and audio is totally gone in some cases. > > How do i troubleshoot this ? And how to tune RTP performance for FreeSwitch. > > I checked that CPU is not fully utilized and is somewhere around 40% range and memory is only at 10%. > > Also network interface is not fully utilized as well as i was able to transfer files from server and interface was able to handle much more bandwidth than it was running. > > Please help. From amkusmirek at gmail.com Mon Nov 24 18:22:32 2014 From: amkusmirek at gmail.com (=?UTF-8?Q?Adam_Ku=C5=9Bmirek?=) Date: Mon, 24 Nov 2014 16:22:32 +0100 Subject: [Freeswitch-users] uuid_send_dtmf problem with rfc2833 Message-ID: Michael, it is not nat problem. Freeswitch is not sending dtmf events. I can't see anything on local interface. Italo, thanks for workaround, i will try it and send feedback. Regards Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/d85b5c35/attachment.html From ssinyagin at gmail.com Mon Nov 24 18:26:11 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Mon, 24 Nov 2014 16:26:11 +0100 Subject: [Freeswitch-users] Fwd: How to provide ringback tone while transferring a parked call In-Reply-To: References: Message-ID: just in case if you're interested, here's a slightly more complex example: https://github.com/xlab1/go-fs-secretary-prototype It accepts the inbound call, plays the music, then makes an outbound call and requests DTMF input, and then bridges them together. So, you have the possibility to do something more with the outbound call, like searching for the user on multiple phones, or finding the free agent in a call center, etc. On Mon, Nov 24, 2014 at 3:01 PM, Szeto, Steven wrote: > Thanks Stanislav & Brian. It worked. > > Here's a summary of the call sequence: > > 1. Originate a call and park it: > > originate user/1001 &park > > > 2. Set the transfer_ringback variable to US ringback: > > uuid_setvar e121f9cb-f9a2-43fa-a61e-2a9a7cd5df00 transfer_ringback > %(2000,4000,440.0,480.0) > > > 3. Now transfer the parked call: > > uuid_transfer e121f9cb-f9a2-43fa-a61e-2a9a7cd5df00 bridge:user/1002 inline > > > > On Mon, Nov 24, 2014 at 4:05 AM, Stanislav Sinyagin > wrote: > >> you just execute >> uuid_setvar [value] >> before executing uuid_transfer >> >> On Mon, Nov 24, 2014 at 2:47 AM, Szeto, Steven >> wrote: >> >>> Thanks, Brian. >>> >>> Can anyone give an example of the syntax to set the transfer_ringback >>> variable within the uuid_transfer command? >>> >>> I've tried a few combos that either result in a syntax error or the call >>> dropping. >>> >>> I think it is supposed to look something like this: >>> >>> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff [ >>> transfer_ringback=$${us-ring}]bridge:user/1002 inline >>> >>> >>> On Sat, Nov 22, 2014 at 10:13 AM, Brian West >>> wrote: >>> >>>> transfer_ringback variable >>>> >>>> >>>> On Friday, November 21, 2014, Szeto, Steven >>>> wrote: >>>> >>>>> >>>>> >>>>> 1. Suppose you originate a call and park it: >>>>> >>>>> originate user/1001 &park >>>>> >>>>> >>>>> 2. Now transfer the parked call: >>>>> >>>>> uuid_transfer d7605992-7e0d-4c24-9634-75fe081422ff bridge:user/1002 >>>>> inline >>>>> >>>>> When you do this, the parked caller hears nothing until the >>>>> destination answers. >>>>> >>>>> How to you pass enough parameters to the uuid_transfer command to play >>>>> ringback tone to the parked call during the transfer? >>>>> >>>>> >>>>> >>>>> This e-mail (including any attachments) is for the sole use of the >>>>> intended recipient(s) and may contain information that is confidential >>>>> and/or protected by legal privilege. Any unauthorized review, use, copy, >>>>> disclosure or distribution of this e-mail is strictly prohibited. If you >>>>> are not the intended recipient, please notify Mitel immediately and destroy >>>>> all copies of this e-mail. Mitel does not accept any liability for breach >>>>> of security, error or virus that may result from the transmission of this >>>>> message. >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >>>> *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> This e-mail (including any attachments) is for the sole use of the >>> intended recipient(s) and may contain information that is confidential >>> and/or protected by legal privilege. Any unauthorized review, use, copy, >>> disclosure or distribution of this e-mail is strictly prohibited. If you >>> are not the intended recipient, please notify Mitel immediately and destroy >>> all copies of this e-mail. Mitel does not accept any liability for breach >>> of security, error or virus that may result from the transmission of this >>> message. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > This e-mail (including any attachments) is for the sole use of the > intended recipient(s) and may contain information that is confidential > and/or protected by legal privilege. Any unauthorized review, use, copy, > disclosure or distribution of this e-mail is strictly prohibited. If you > are not the intended recipient, please notify Mitel immediately and destroy > all copies of this e-mail. Mitel does not accept any liability for breach > of security, error or virus that may result from the transmission of this > message. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/55dc88e7/attachment.html From rtreleaven at bunnykick.ca Mon Nov 24 18:30:46 2014 From: rtreleaven at bunnykick.ca (Russell Treleaven) Date: Mon, 24 Nov 2014 10:30:46 -0500 Subject: [Freeswitch-users] RTP Packet Loss Troubleshooting In-Reply-To: <184C1B38-47AE-4BB2-89F2-100C5EB71330@jerris.com> References: <184C1B38-47AE-4BB2-89F2-100C5EB71330@jerris.com> Message-ID: check for dropped packets on the interfaces On Mon, Nov 24, 2014 at 10:00 AM, Michael Jerris wrote: > Are you sure you are not on a 100mb or 1/2 duplex link in your network > path? These sorts of issues typically get down to network hardware, > network drivers, os cache settings, and that sort of thing. Lots of > devices and hardware do not care for high packets per second of voip. This > could just be a junk switch in the mix. > > > > On Oct 16, 2014, at 8:39 AM, Linux Vince wrote: > > > > I have installed freeswitch and testing with SiPP > > > > FreeSwitch was easily able to handle 1700 concurrent calls. > > > > However when it starts going near 2000 my RTP packets are being lost and > audio is totally gone in some cases. > > > > How do i troubleshoot this ? And how to tune RTP performance for > FreeSwitch. > > > > I checked that CPU is not fully utilized and is somewhere around 40% > range and memory is only at 10%. > > > > Also network interface is not fully utilized as well as i was able to > transfer files from server and interface was able to handle much more > bandwidth than it was running. > > > > Please help. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/229d8474/attachment-0001.html From danb.lists at gmail.com Mon Nov 24 18:32:16 2014 From: danb.lists at gmail.com (Dan Christian Bogos) Date: Mon, 24 Nov 2014 16:32:16 +0100 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: References: Message-ID: <54734F80.3060802@gmail.com> Guys, Many thanks for your answers. @Vik: Yes, I am having the xml_curl correctly configured (I provision almost all, dialplan, directory, config, phrases in this way, and no issues elsewhere). @Steven: I thought I do. If I use exactly the same configuration received from xml_curl in fifo.conf, I have no issues. Bellow you can find what I am returning to FS, maybe you can spot some error or namespace issue? @Michael: I am loading mod_xml_curl very early at start (right after mod_console and mod_syslog) and mod_fifo way later. I thought that if I ask for a sample maybe I can spot the mistake (easy one I have imagined it) myself. Bellow you can find the configuration retrieval when I use console command "reload mod_fifo" Thanks again! DanB """ T 2014/11/24 16:22:18.818134 192.168.50.136:53681 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 107. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf ## T 2014/11/24 16:22:18.818764 192.168.50.136:2880 -> 192.168.50.136:53681 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. .
###### T 2014/11/24 16:22:18.819212 192.168.50.136:53682 -> 192.168.50.136:2880 [AP] POST /phrases/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 107. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf ## T 2014/11/24 16:22:18.819931 192.168.50.136:2880 -> 192.168.50.136:53682 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. .
###### T 2014/11/24 16:22:18.820574 192.168.50.136:53683 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 109. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf ## T 2014/11/24 16:22:18.820957 192.168.50.136:2880 -> 192.168.50.136:53683 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:18 GMT. Content-Length: 105. Content-Type: text/plain; charset=utf-8. .
###### T 2014/11/24 16:22:18.821345 192.168.50.136:53684 -> 192.168.50.136:2880 [AP] POST /phrases/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 109. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf T 2014/11/24 16:22:19.517231 192.168.50.136:53685 -> 192.168.50.136:2880 [AP] POST /config/ HTTP/1.1. User-Agent: freeswitch-xml/1.0. Host: 192.168.50.136:2880. Accept: */*. Content-Length: 104. Content-Type: application/x-www-form-urlencoded. . hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=fifo.conf ## T 2014/11/24 16:22:19.520502 192.168.50.136:2880 -> 192.168.50.136:53685 [AP] HTTP/1.1 200 OK. Date: Mon, 24 Nov 2014 15:22:19 GMT. Content-Length: 880. Content-Type: text/plain; charset=utf-8. . {sip_invite_domain=sip.test.domain.com,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u4_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u4_0_EPR at sip.test.domain.com;fs_path=sip:127.0.0.1{sip_invite_domain=sip.test.domain.com,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u205_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u205_0_EPR at sip.test.domain.com;fs_path=sip:127.0.0.1 """ From vipkilla at gmail.com Mon Nov 24 18:45:13 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 24 Nov 2014 10:45:13 -0500 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: <54734F80.3060802@gmail.com> References: <54734F80.3060802@gmail.com> Message-ID: You maybe missing the section tag, see below:
...
On Mon, Nov 24, 2014 at 10:32 AM, Dan Christian Bogos wrote: > Guys, > > Many thanks for your answers. > > @Vik: Yes, I am having the xml_curl correctly configured (I provision > almost all, dialplan, directory, config, phrases in this way, and no > issues elsewhere). > @Steven: I thought I do. If I use exactly the same configuration > received from xml_curl in fifo.conf, I have no issues. Bellow you can > find what I am returning to FS, maybe you can spot some error or > namespace issue? > @Michael: I am loading mod_xml_curl very early at start (right after > mod_console and mod_syslog) and mod_fifo way later. > > I thought that if I ask for a sample maybe I can spot the mistake (easy > one I have imagined it) myself. > > Bellow you can find the configuration retrieval when I use console > command "reload mod_fifo" > > > Thanks again! > > DanB > > """ > > T 2014/11/24 16:22:18.818134 192.168.50.136:53681 -> 192.168.50.136:2880 > [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880. > Accept: */*. > Content-Length: 107. > Content-Type: application/x-www-form-urlencoded. > . > > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf > ## > T 2014/11/24 16:22:18.818764 192.168.50.136:2880 -> 192.168.50.136:53681 > [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.819212 192.168.50.136:53682 -> 192.168.50.136:2880 > [AP] > POST /phrases/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880. > Accept: */*. > Content-Length: 107. > Content-Type: application/x-www-form-urlencoded. > . > > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf > ## > T 2014/11/24 16:22:18.819931 192.168.50.136:2880 -> 192.168.50.136:53682 > [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.820574 192.168.50.136:53683 -> 192.168.50.136:2880 > [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880. > Accept: */*. > Content-Length: 109. > Content-Type: application/x-www-form-urlencoded. > . > > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf > ## > T 2014/11/24 16:22:18.820957 192.168.50.136:2880 -> 192.168.50.136:53683 > [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.821345 192.168.50.136:53684 -> 192.168.50.136:2880 > [AP] > POST /phrases/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880. > Accept: */*. > Content-Length: 109. > Content-Type: application/x-www-form-urlencoded. > . > > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf > > > T 2014/11/24 16:22:19.517231 192.168.50.136:53685 -> 192.168.50.136:2880 > [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880. > Accept: */*. > Content-Length: 104. > Content-Type: application/x-www-form-urlencoded. > . > > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=fifo.conf > ## > T 2014/11/24 16:22:19.520502 192.168.50.136:2880 -> 192.168.50.136:53685 > [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:19 GMT. > Content-Length: 880. > Content-Type: text/plain; charset=utf-8. > . > importance="0">{sip_invite_domain= > sip.test.domain.com,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag= > sip.test.domain.com > ,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u4_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/ > eprou-a41_u4_0_EPR at sip.test.domain.com;fs_path=sip:127.0.0.1 timeout="30" simo="1" lag="0">{sip_invite_domain=sip.test.domain.com > ,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com > ,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u205_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/ > eprou-a41_u205_0_EPR at sip.test.domain.com > ;fs_path=sip:127.0.0.1 > > """ > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/66b668d4/attachment.html From mike at jerris.com Mon Nov 24 18:51:57 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 10:51:57 -0500 Subject: [Freeswitch-users] uuid_send_dtmf problem with rfc2833 In-Reply-To: References: Message-ID: <84A8D55D-87A1-4879-8EED-52160F79A1B1@jerris.com> Please file a jira (even better when including a pull request to fix it :D ) > On Nov 24, 2014, at 10:22 AM, Adam Ku?mirek wrote: > > Michael, it is not nat problem. Freeswitch is not sending dtmf events. I can't see anything on local interface. > > Italo, thanks for workaround, i will try it and send feedback. > > Regards Adam > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dsaraginov at gmail.com Mon Nov 24 18:46:26 2014 From: dsaraginov at gmail.com (Dragan Saraginov) Date: Mon, 24 Nov 2014 16:46:26 +0100 Subject: [Freeswitch-users] Freeswitch and virtual FAX Message-ID: Hi guys, I hope someone can help me on the issue that I am facing. I am planing to implement a virtual FAX machine with freeswitch. My main concern is, will it be possible to do that without any type of card, analog, digital, ATA gateway or IAXmodem, but using the t38 support? Best regards, Dragan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/41da7e92/attachment-0001.html From mike at jerris.com Mon Nov 24 18:54:52 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 10:54:52 -0500 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: References: <54734F80.3060802@gmail.com> Message-ID: <61D732F5-DC7A-4D24-930B-3EAB48A91D33@jerris.com> don't load mod_fifo on startup at fs_cli do: xml_curl debug_on load mod_fifo does it even try to make the request? If not, double check your xml_curl config bindings for configuration. > On Nov 24, 2014, at 10:45 AM, Vik Killa wrote: > > You maybe missing the section tag, see below: > > >
> > ... > >
> > > On Mon, Nov 24, 2014 at 10:32 AM, Dan Christian Bogos > wrote: > Guys, > > Many thanks for your answers. > > @Vik: Yes, I am having the xml_curl correctly configured (I provision > almost all, dialplan, directory, config, phrases in this way, and no > issues elsewhere). > @Steven: I thought I do. If I use exactly the same configuration > received from xml_curl in fifo.conf, I have no issues. Bellow you can > find what I am returning to FS, maybe you can spot some error or > namespace issue? > @Michael: I am loading mod_xml_curl very early at start (right after > mod_console and mod_syslog) and mod_fifo way later. > > I thought that if I ask for a sample maybe I can spot the mistake (easy > one I have imagined it) myself. > > Bellow you can find the configuration retrieval when I use console > command "reload mod_fifo" > > > Thanks again! > > DanB > > """ > > T 2014/11/24 16:22:18.818134 192.168.50.136:53681 -> 192.168.50.136:2880 [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880 . > Accept: */*. > Content-Length: 107. > Content-Type: application/x-www-form-urlencoded. > . > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf > ## > T 2014/11/24 16:22:18.818764 192.168.50.136:2880 -> 192.168.50.136:53681 [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.819212 192.168.50.136:53682 -> 192.168.50.136:2880 [AP] > POST /phrases/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880 . > Accept: */*. > Content-Length: 107. > Content-Type: application/x-www-form-urlencoded. > . > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=spandsp.conf > ## > T 2014/11/24 16:22:18.819931 192.168.50.136:2880 -> 192.168.50.136:53682 [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.820574 192.168.50.136:53683 -> 192.168.50.136:2880 [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880 . > Accept: */*. > Content-Length: 109. > Content-Type: application/x-www-form-urlencoded. > . > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf > ## > T 2014/11/24 16:22:18.820957 192.168.50.136:2880 -> 192.168.50.136:53683 [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:18 GMT. > Content-Length: 105. > Content-Type: text/plain; charset=utf-8. > . >
> ###### > T 2014/11/24 16:22:18.821345 192.168.50.136:53684 -> 192.168.50.136:2880 [AP] > POST /phrases/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880 . > Accept: */*. > Content-Length: 109. > Content-Type: application/x-www-form-urlencoded. > . > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=timezones.conf > > > T 2014/11/24 16:22:19.517231 192.168.50.136:53685 -> 192.168.50.136:2880 [AP] > POST /config/ HTTP/1.1. > User-Agent: freeswitch-xml/1.0. > Host: 192.168.50.136:2880 . > Accept: */*. > Content-Length: 104. > Content-Type: application/x-www-form-urlencoded. > . > hostname=nl-asd-dev-sbc01§ion=configuration&tag_name=configuration&key_name=name&key_value=fifo.conf > ## > T 2014/11/24 16:22:19.520502 192.168.50.136:2880 -> 192.168.50.136:53685 [AP] > HTTP/1.1 200 OK. > Date: Mon, 24 Nov 2014 15:22:19 GMT. > Content-Length: 880. > Content-Type: text/plain; charset=utf-8. > . > {sip_invite_domain=sip.test.domain.com ,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com ,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u4_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u4_0_EPR at sip.test.domain.com ;fs_path=sip:127.0.0.1{sip_invite_domain=sip.test.domain.com ,sip_h_X-AuthType=INTERNAL,sip_h_X-DomainTag=sip.test.domain.com ,sip_h_X-EpType=SIP,sip_h_X-EpTag=endpoint2,sip_h_X-Destination=a41_u205_0_EPR,sip_h_X-Balancer=1.2.3.4,call_timeout=30}sofia/ipbxas/eprou-a41_u205_0_EPR at sip.test.domain.com ;fs_path=sip:127.0.0.1 > > """ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/7d76fd6e/attachment.html From mike at jerris.com Mon Nov 24 18:55:51 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 10:55:51 -0500 Subject: [Freeswitch-users] Freeswitch and virtual FAX In-Reply-To: References: Message-ID: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> assuming you have a provider that supports t.38, sure. > On Nov 24, 2014, at 10:46 AM, Dragan Saraginov wrote: > > Hi guys, > > I hope someone can help me on the issue that I am facing. I am planing to implement a virtual FAX machine with freeswitch. My main concern is, will it be possible to do that without any type of card, analog, digital, ATA gateway or IAXmodem, but using the t38 support? From bordmi at rarus.ru Mon Nov 24 19:03:11 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Mon, 24 Nov 2014 20:03:11 +0400 Subject: [Freeswitch-users] Server death bug on call origination Message-ID: Hi, All! When I do next: Dialplan: sofia/internal/104 at company1.XXX.com Action answer() Dialplan: sofia/internal/104 at company1.XXX.com Action bridge([presence_id= 101 at company1.XXX.com]sofia/internal/sip:101 at XXX.XXX.XXX.98 :1966;rinstance=91c49290308619e2:_:[presence_id=102 at company1.XXX.com ]sofia/internal/sip:102 at XXX.XXX.XXX.98:15582;rinstance=d04d400035af5525) my server is died with 100% CPU load and I will able recover it only by hardware reset. trying this on v1.4.9 and master current bordmi at switch:~ % uname -a FreeBSD switch.XXX.com 10.0-RELEASE-p10 FreeBSD 10.0-RELEASE-p10 #0: Mon Oct 20 12:42:25 UTC 2014 root at amd64-builder.daemonology.net:/usr/obj/usr/src/sys/GENERIC amd64 freeswitch at internal> version FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit) Virtual hardware by VMWare vSphere -- Regards, Dmitriy Borisov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/44b36a36/attachment.html From mike at jerris.com Mon Nov 24 19:29:54 2014 From: mike at jerris.com (Michael Jerris) Date: Mon, 24 Nov 2014 11:29:54 -0500 Subject: [Freeswitch-users] Server death bug on call origination In-Reply-To: References: Message-ID: <4483D4E0-0587-42C8-A7F8-17C86E84A726@jerris.com> https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > On Nov 24, 2014, at 11:03 AM, ???????, ??????? wrote: > > Hi, All! > > When I do next: > Dialplan: sofia/internal/104 at company1.XXX.com Action answer() > Dialplan: sofia/internal/104 at company1.XXX.com Action bridge([presence_id=101 at company1.XXX.com ]sofia/internal/sip:101 at XXX.XXX.XXX.98:1966;rinstance=91c49290308619e2:_:[presence_id=102 at company1.XXX.com ]sofia/internal/sip:102 at XXX.XXX.XXX.98:15582;rinstance=d04d400035af5525) > my server is died with 100% CPU load and I will able recover it only by hardware reset. > > trying this on v1.4.9 and master current > > bordmi at switch:~ % uname -a > FreeBSD switch.XXX.com 10.0-RELEASE-p10 FreeBSD 10.0-RELEASE-p10 #0: Mon Oct 20 12:42:25 UTC 2014 root at amd64-builder.daemonology.net:/usr/obj/usr/src/sys/GENERIC amd64 > > freeswitch at internal> version > FreeSWITCH Version 1.4.9+git~20140929T194948Z~ae069dcca7~64bit (git ae069dc 2014-09-29 19:49:48Z 64bit) > > Virtual hardware by VMWare vSphere -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/556b3919/attachment-0001.html From danb.lists at gmail.com Mon Nov 24 21:48:50 2014 From: danb.lists at gmail.com (Dan Christian Bogos) Date: Mon, 24 Nov 2014 19:48:50 +0100 Subject: [Freeswitch-users] Provisioning fifo.conf via xml_curl In-Reply-To: References: Message-ID: <54737D92.1030208@gmail.com> Guys, Again thanks for your quick support, worked this time! @Vik: that was it, t and
was missing out of the returned configuration. I believed somehow that fifo.conf is different than the rest of the stuff I was provisioning by directly copying the fifo.conf to be provisioned, so that was my mistake. @Michael: It works now with load in modules.conf as well as without fifo.conf file being in autoload_configs, so problem solved! Beer on me :). Cheers, DanB From vipkilla at gmail.com Mon Nov 24 21:55:39 2014 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 24 Nov 2014 13:55:39 -0500 Subject: [Freeswitch-users] Freeswitch and virtual FAX In-Reply-To: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> References: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> Message-ID: We have a PRI on a Cisco VVX that converts it to SIP and sends to FS and spandsp receives the fax. We enabled T38 on the VVX by default but it doesn't always work, for some numbers have use straight g711 ulaw which seems to work too. On Mon, Nov 24, 2014 at 10:55 AM, Michael Jerris wrote: > assuming you have a provider that supports t.38, sure. > > > On Nov 24, 2014, at 10:46 AM, Dragan Saraginov > wrote: > > > > Hi guys, > > > > I hope someone can help me on the issue that I am facing. I am planing > to implement a virtual FAX machine with freeswitch. My main concern is, > will it be possible to do that without any type of card, analog, digital, > ATA gateway or IAXmodem, but using the t38 support? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/c8c0ede6/attachment.html From krice at freeswitch.org Mon Nov 24 23:11:42 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 24 Nov 2014 20:11:42 +0000 Subject: [Freeswitch-users] Freeswitch Week in Review (Master Branch) November 16th-22nd Message-ID: <547390fe44beb_74526b932479485@ip-10-142-130-179.mail> New Post on freeswitch.org from kathleen check it out at http://ift.tt/1FkkF5y Freeswitch Week in Review (Master Branch) November 16th-22nd Hello, again. This week in the FreeSWITCH master branch we had 25 commits. The new features for this week are: the addition of `connect-timeout` option for curl API command, allow setting CURL timeout from curl API command , respecting unmasked when creating new files in mod_sndfile , the addition of ignore_connect_fail config setting (to not kill the call when redis is down when using redis back-end for limit), support for additional curl auth methods in mod_curl , fifo_position macro, improvements to timerfd implementation (to be more accurate) , and mod_rayo to no longer support SSLv2/3 due to security concerns. New features that were added: 1ee325d Add `connect-timeout` option for curl API command f1df8d6 Allow setting CURL timeout from curl API command c73afe1 FS-7004 Respect unmasked when creating new files in mod_sndfile [Jira: http://ift.tt/1pgbVcN] 6f660c3 Updated mod_rayo to no longer support SSLv2/3 f198d82 FS-5666 Add ignore_connect_fail config setting to not kill the call when redis is down when using redis back-end for limit [Jira: http://ift.tt/1pgbVcP] 250234d FS-5800 Add support for additional curl auth methods in mod_curl [Jira: http://ift.tt/1pgbVcR] 1e92619 FS-6097 ? added fifo_position macro [Jira: http://ift.tt/1FkkFCz] da6043f Improve timerfd implementation to be more accurate df423b8 Improve timerfd implementation to be more accurate Improvements in cross platform build supports: 9673cf0 Fix for apr build issue fs-6848 freebsd arm 62a2e10 Remove hack breaking some cross compile builds. if you really need this, you should be using new enough glibc anyways. The following bugs were squashed: 0cf770a FS-6996: #resolve fix define change as of glibc 2.20 for _BSD_SOURCE -> _DEFAULT_SOURCE [Jira: http://ift.tt/1pgbVcT] 424df19 FS-Fix for build failure on mips[Jira: http://ift.tt/1pgbU8z] 8eaaa08 FS-6622 Set buffer size for streams based on the number of channels to avoid buffer starvation in mod_shout [Jira: http://ift.tt/1pgbVtb] 5127b64 FS-7014 Don?t touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt in mod_sofia [Jira: http://ift.tt/1FkkFCF] 8330336 FS-6450 Update library [apr] Backport APR_RING_FOREACH and APR_RING_FOREACH_SAFE macros to APR for unimrcp compatibility.[Jira: http://ift.tt/1pgbVtf] c645ab3 FS-6848 #resolve #comment slight tweak to fix the logic and avoid unbalanced parens [Jira: http://ift.tt/1FkkE1B] 79de78a FS-7021 Remove extra loops when using originate_retries and the caller hangs up.[Jira: http://ift.tt/1pgbUp3] The complete list of commits can be found here:2014_11_16-2014_11_23 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/9451ad06/attachment.html From krice at freeswitch.org Tue Nov 25 00:27:13 2014 From: krice at freeswitch.org (Ken Rice) Date: Mon, 24 Nov 2014 21:27:13 +0000 Subject: [Freeswitch-users] FreeSWITCH 1.4.14 Released Message-ID: <5473a2b195f0d_ea61b9732884134@ip-10-156-208-135.mail> New Post on freeswitch.org from krice387 check it out at http://ift.tt/1zmebiE FreeSWITCH 1.4.14 Released FreeSWITCH 1.4.14 has been released! This is routine maintenance release. Quick highlight of improvements: support for logging full timestamps with dialplan (defaults to old behavior unless requested), allow mod_http_cache to S3 services other than Amazon, and new configuration parameter add-variables-to-offer, addition of `connect-timeout` option for curl API command, allow setting CURL timeout from curl API command , respecting umask when creating new files in mod_sndfile , support for additional curl auth methods in mod_curl , fifo_position macro, improvement of accuracy in timerfd implementation , and remove support SSLv2/3 from mod_rayo due to security concerns. Tarballs and Packages available at the usual locations. New features that were added: 7c0cf50 Set managedlist to return the value to the API stream and the log in mod_managed 4037e78 Support per-module references directory in mod_managed 10ebeba Add console log level in mod_managed 33cb950 Added pure CreateStateHandlerDelegate in ManagedSession for native api usage in mod_managed 889b678 Added GetPtr to Util class for internal pointers extraction (very useful when using native api) in mod_managed 8478874 FS-6831 Addition of unset app for the chatplan and while you can already unset by calling set with var=, same with set in dialplan, this is a convenience function similar to our unset in dialplan in mod_sms [Jira: http://ift.tt/1Hwa0Xw] bc767bb Adding rfc6598.auto and adding rfc6598 space to nat.auto acl. This is the NAT444 carrier grade nat space. 1190e59 FS-6965 Set the Accept-Language into a variable when received in mod_sofia [Jira: http://ift.tt/1uLG7M3] 94278b5 FS-5159 Allow enter and exit sounds to interrupt the MOH in a wait_mod conference 1944f9a FS-6968 Changes to mod_fifo.c to add outbound_per_cycle_min in mod_fifo [Jira: http://ift.tt/1Hwa0XA] 0f2816d Add command to compile non-minified js file for testing in mod_verto f175c71 FS-6805 add support for logging full timestamps with dialplan, defaults to old behavior unless requested. Jira: http://ift.tt/11E7kpC fada4b8 FS-6977 Do not create freeswitch.serial if zrtp not enabled Jira: http://ift.tt/11E7kpG dd629c1 Add external_video_source to media handle and expose switch_core_media_start_video_thread() to start the core video thread for non-rtp based media 0eefdca FS-6947 Opus RTP payload fmtp settings ( maxaveragebitrate / maxplaybackrate ) Jira: http://ift.tt/11M58xc dd61232 FS-6979 Allow mod_http_cache to S3 services other than Amazon Jira: http://ift.tt/11E7i0R 826d428 FS-6992 Global configuration or maxplaybackrate and maxaveragebitrate from opus.conf.xml Jira: http://ift.tt/11M58xe e1c0ef5 New configuration parameter, add-variables-to-offer (default=false) (When true, all channel variables are included in the offer to rayo client in mod_rayo). 1ee325d Add `connect-timeout` option for curl API command f1df8d6 Allow setting CURL timeout from curl API command c73afe1 FS-7004 Respect unmasked when creating new files in mod_sndfile [Jira: http://ift.tt/1pgbVcN] 6f660c3 Updated mod_rayo to no longer support SSLv2/3 f198d82 FS-5666 Add ignore_connect_fail config setting to not kill the call when redis is down when using redis back-end for limit [Jira: http://ift.tt/1pgbVcP] 250234d FS-5800 Add support for additional curl auth methods in mod_curl [Jira: http://ift.tt/1pgbVcR] 1e92619 FS-6097 ? added fifo_position macro [Jira: http://ift.tt/1FkkFCz] da6043f Improve timerfd implementation to be more accurate df423b8 Improve timerfd implementation to be more accurate Improvements in cross platform build supports: 0f8b993 Fix mod_say_es_ar Makefile.am 07030c6 Fix compiler warning on unmatched return type 5dee5ce FS-6953 Fix for build issues with OS X [Jira: http://ift.tt/1uLG9Um] a17be38 Add reconf target 300b8d8 FS-6973 Fix for changed Opal svn url breaking build of mod_opal [Jira: http://ift.tt/1Hwa1dY] 9673cf0 Fix for apr build issue FS-6848 freebsd arm 62a2e10 Remove hack breaking some cross compile builds. if you really need this, you should be using new enough glibc anyways. The following bugs were squashed: f87c335 Fix for issue with tracking agent availability 8f3c157 FS-6957 Fix for muting issues upon joining a conference in mod_conference [Jira: http://ift.tt/1uLG9Ue] 831832c FS-6890 Fix for FreeSWITCH forgets to send RE-INVITES [Jira: http://ift.tt/1uLG9Ug] 4eb5b38 Fix bug where re-invites needlessly re-init the codec and jb a497169 FS-6890 Additional work for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 9c1e603 FS-6954 #comment Fix for side effect of another fix for this bug. Please test this patch. [Jira: http://ift.tt/1yzQC5V] f66f2ca FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia [Jira: http://ift.tt/1uLG9Ug] 415f82f FS-6954 Work toward fix for Freeswitch adding additional m=audio on t.38 call in mod_sofia [Jira: http://ift.tt/1yzQC5V] 5ce5199 FS-6969 Fix for unwanted profile shutdown in mod_verto [Jira: http://ift.tt/1Hwa0XE] In terms of stability these were the use cases that were fixed: f3d089a Fix a crash when trying to remove a shadow directory in mod_managed 2f1b12f OPENZAP-232 Check for digits received on sangoma isdn stack to avoid delaying moving to the ring state if all digits are received at once in overlap dialing mode. 6b8d5b2 Fix release guard timer check in freetdm 34cf3b9 FS-6980 #resolve Fix for crash when using native recording on recordstop Jira: http://ift.tt/11E7i0U ab24bde FS-5533 fix issue with busy signal being sent back to all shared lines instead of just the calling device Jira: http://ift.tt/11M58xi a279bf3 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 0c68bb6 FS-6957 Fix for muting issues in mod_conference Jira: http://ift.tt/1uLG9Ue 33d37ce PLIV-13 Disable SSLv3 in libcurl for mod_httapi dbc5571 FS-6983 Wrap new curl TLS macro usage with ifdefs Jira: http://ift.tt/11E7i0W 0699ea8 Fixes for various issues 87a4670 FS-6890 More work toward fix for FreeSWITCH forgetting to send RE-INVITES in mod_sofia Jira: http://ift.tt/1uLG9Ug 82aa331 FS-6531: #resolve set to tag on auto answer notify Jira: http://ift.tt/11M58xk 75473a7 FS-6531: #resolve set to tag on uuid_phone_event notify Jira: http://ift.tt/11M58xk 07c5cc1 FS-7003 Fix infinite loop when output sent to server without SSML configured and repeat-times=0 Jira: http://ift.tt/11E7kG0 0cf770a FS-6996: #resolve fix define change as of glibc 2.20 for _BSD_SOURCE -> _DEFAULT_SOURCE [Jira: http://ift.tt/1pgbVcT] 424df19 FS-Fix for build failure on mips[Jira: http://ift.tt/1pgbU8z] 8eaaa08 FS-6622 Set buffer size for streams based on the number of channels to avoid buffer starvation in mod_shout [Jira: http://ift.tt/1pgbVtb] 5127b64 FS-7014 Don?t touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt in mod_sofia [Jira: http://ift.tt/1FkkFCF] 8330336 FS-6450 Update library [apr] Backport APR_RING_FOREACH and APR_RING_FOREACH_SAFE macros to APR for unimrcp compatibility.[Jira: http://ift.tt/1pgbVtf] c645ab3 FS-6848 #resolve #comment slight tweak to fix the logic and avoid unbalanced parens [Jira: http://ift.tt/1FkkE1B] 79de78a FS-7021 Remove extra loops when using originate_retries and the caller hangs up.[Jira: http://ift.tt/1pgbUp3] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141124/b3dee8dd/attachment-0001.html From youssef.elouam at outlook.com Tue Nov 25 12:02:55 2014 From: youssef.elouam at outlook.com (youssef elouam) Date: Tue, 25 Nov 2014 09:02:55 +0000 Subject: [Freeswitch-users] mod_sofia.c:2219 CODEC NEGOTIATION ERROR Message-ID: Dear; Many thanks for your reply. I have upgraded gcc to 4.8 but i still have problem. I will reinstall freeswitch from the beginning and try again. Youssef ELOUAM --- Original Message --- From: "Ken Rice" Sent: 23 November 2014 19:19 To: "FreeSWITCH Users Help" Subject: Re: [Freeswitch-users] mod_sofia.c:2219 CODEC NEGOTIATION ERROR You need to be a bit more specific then that.. That?s only showing the SDP... But on the Raspberry PI there is a known GCC bug if you are using an older version of GCC see http://lists.freeswitch.org/pipermail/freeswitch-users/2014-June/105974.html On 11/23/14 10:58 AM, "youssef elouam" wrote: > Dear All ; > > many thanks in advance for your help. I have problem of CODEC NEGOTIATION in > my freeswitch installed on Raspberry PI board. > > following is logs I have : > > 2014-11-23 16:52:16.832177 [ERR] mod_sofia.c:2219 CODEC NEGOTIATION ERROR. > SDP: > v=0 > o=- 3625750336 3625750336 IN IP4 192.168.1.5 > s=pjmedia > c=IN IP4 192.168.1.5 > t=0 0 > m=audio 4006 RTP/AVP 99 0 8 101 > c=IN IP4 192.168.1.5 > a=rtpmap:99 SILK/24000 > a=fmtp:99 useinbandfec=0 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtcp:4007 IN IP4 192.168.1.5 > > > freeswitch at internal> version > FreeSWITCH Version 1.5.15b+git~20141024T202644Z~12b6940644~32bit (git 12b6940 > 2014-10-24 20:26:44Z 32bit) > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141125/f6972241/attachment.html -------------- next part -------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From aqsyounas at gmail.com Tue Nov 25 22:57:54 2014 From: aqsyounas at gmail.com (Aqs Younas) Date: Wed, 26 Nov 2014 00:57:54 +0500 Subject: [Freeswitch-users] Can i play i media while shout is fetching url Message-ID: Hi, Users Can i play media while my mod_shout is fetching url stream from the server. Like, "Please wait we are getting your requested url" I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/6f7eb209/attachment.html From mario_fs at mgtech.com Wed Nov 26 00:21:37 2014 From: mario_fs at mgtech.com (Mario G) Date: Tue, 25 Nov 2014 13:21:37 -0800 Subject: [Freeswitch-users] jira question about status change Message-ID: I provided info to an open jira (FS-7027) which had a ?waiting for info? status, I then tried to change the status but messed it up, apparently the supplier cannot change it back to open. When a jira is in this status and the info is fulfilled, how does the status get updated that the info was fulfilled? Mario G From mike at jerris.com Wed Nov 26 00:31:00 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Nov 2014 16:31:00 -0500 Subject: [Freeswitch-users] Can i play i media while shout is fetching url In-Reply-To: References: Message-ID: <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> This sounds like a cool feature, but its not something that you can do right now. I'd be happy to review a patch to add this. > On Nov 25, 2014, at 2:57 PM, Aqs Younas wrote: > > Hi, Users > > Can i play media while my mod_shout is fetching url stream from the server. > > Like, "Please wait we are getting your requested url" > > I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . From mike at jerris.com Wed Nov 26 00:32:04 2014 From: mike at jerris.com (Michael Jerris) Date: Tue, 25 Nov 2014 16:32:04 -0500 Subject: [Freeswitch-users] jira question about status change In-Reply-To: References: Message-ID: <75F5C04B-D0E0-4BC1-A871-63F202DFB16D@jerris.com> we were actually just now discussing our upcoming workflow changes. I hope this gets addressed soon. > On Nov 25, 2014, at 4:21 PM, Mario G wrote: > > I provided info to an open jira (FS-7027) which had a ?waiting for info? status, I then tried to change the status but messed it up, apparently the supplier cannot change it back to open. When a jira is in this status and the info is fulfilled, how does the status get updated that the info was fulfilled? > Mario G From martin.hoole at emailn.de Wed Nov 26 03:06:46 2014 From: martin.hoole at emailn.de (martin.hoole at emailn.de) Date: Wed, 26 Nov 2014 01:06:46 +0100 Subject: [Freeswitch-users] VAD / Energy level of talking mod_verto Message-ID: <2124afdd99eb84da2953a6b468dd20ba@mail.emailn.de> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/dc66f98f/attachment.html From anthony.minessale at gmail.com Wed Nov 26 03:39:16 2014 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 25 Nov 2014 18:39:16 -0600 Subject: [Freeswitch-users] VAD / Energy level of talking mod_verto In-Reply-To: <2124afdd99eb84da2953a6b468dd20ba@mail.emailn.de> References: <2124afdd99eb84da2953a6b468dd20ba@mail.emailn.de> Message-ID: Its probably possible but not implemented. Would require a bounty. On Tue, Nov 25, 2014 at 6:06 PM, wrote: > Does mod_verto and mod_conference support VAD / Energy level of talking. > > Currently FS pushes "Talking" , "Floor" , etc > > I would like to have an event between 0-1 for the verbosity of someone > talking? > > > ------------------------------ > > Versendet mit Emailn.de - Freemail > > * Unbegrenzt Speicherplatz > * Eigenes Online-B?ro > * 24h besten Mailempfang > * Spamschutz, Adressbuch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II ? @anthmfs ? @FreeSWITCH ? ? http://freeswitch.org/ ? http://cluecon.com/ ? http://twitter.com/FreeSWITCH ? irc.freenode.net #freeswitch ? *http://freeswitch.org/g+ * ClueCon Weekly Development Call ? sip:888 at conference.freeswitch.org ? +19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141125/045bf215/attachment-0001.html From manish.talwar at nexxuspg.com Wed Nov 26 08:12:09 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Wed, 26 Nov 2014 05:12:09 +0000 Subject: [Freeswitch-users] Can i play i media while shout is fetching url In-Reply-To: <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> References: , <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> Message-ID: <1417027382189.69016@nexxuspg.com> Hi, We are also looking for same feature to use in our application. We want to play a sound like "please wait..." for those cases where it can take around 10 seconds to get a response. This feature really sound good and must be included in FreeSwitch. Please let us know whenever this feature will be included in FreeSwitch. Thanks, Regards, Manish Talwar ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Jerris Sent: 26 November 2014 03:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url This sounds like a cool feature, but its not something that you can do right now. I'd be happy to review a patch to add this. > On Nov 25, 2014, at 2:57 PM, Aqs Younas wrote: > > Hi, Users > > Can i play media while my mod_shout is fetching url stream from the server. > > Like, "Please wait we are getting your requested url" > > I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ssinyagin at gmail.com Wed Nov 26 15:37:18 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 26 Nov 2014 13:37:18 +0100 Subject: [Freeswitch-users] Can i play i media while shout is fetching url In-Reply-To: <1417027382189.69016@nexxuspg.com> References: <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> <1417027382189.69016@nexxuspg.com> Message-ID: you can make a workaround: with an external script, fetch the whole audio file, and play the "please wait" while it's getting downloaded, and then play the audio. Should be quite easy if you use ESL and multithreading. On Wed, Nov 26, 2014 at 6:12 AM, Manish Talwar wrote: > Hi, > > We are also looking for same feature to use in our application. We want to play a sound like "please wait..." for those cases where it can take around 10 seconds to get a response. This feature really sound good and must be included in FreeSwitch. > > Please let us know whenever this feature will be included in FreeSwitch. > > Thanks, > > Regards, > Manish Talwar > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Jerris > Sent: 26 November 2014 03:01 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url > > This sounds like a cool feature, but its not something that you can do right now. I'd be happy to review a patch to add this. > >> On Nov 25, 2014, at 2:57 PM, Aqs Younas wrote: >> >> Hi, Users >> >> Can i play media while my mod_shout is fetching url stream from the server. >> >> Like, "Please wait we are getting your requested url" >> >> I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From manish.talwar at nexxuspg.com Wed Nov 26 15:47:51 2014 From: manish.talwar at nexxuspg.com (Manish Talwar) Date: Wed, 26 Nov 2014 12:47:51 +0000 Subject: [Freeswitch-users] Can i play i media while shout is fetching url In-Reply-To: References: <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> <1417027382189.69016@nexxuspg.com>, Message-ID: <1417054723570.27045@nexxuspg.com> Its not like we always want to play "please wait" audio file, but we need to play this file only for those cases where it can take around long time to get a response. It will be depend on response time only. ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Stanislav Sinyagin Sent: 26 November 2014 18:07 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url you can make a workaround: with an external script, fetch the whole audio file, and play the "please wait" while it's getting downloaded, and then play the audio. Should be quite easy if you use ESL and multithreading. On Wed, Nov 26, 2014 at 6:12 AM, Manish Talwar wrote: > Hi, > > We are also looking for same feature to use in our application. We want to play a sound like "please wait..." for those cases where it can take around 10 seconds to get a response. This feature really sound good and must be included in FreeSwitch. > > Please let us know whenever this feature will be included in FreeSwitch. > > Thanks, > > Regards, > Manish Talwar > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Jerris > Sent: 26 November 2014 03:01 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url > > This sounds like a cool feature, but its not something that you can do right now. I'd be happy to review a patch to add this. > >> On Nov 25, 2014, at 2:57 PM, Aqs Younas wrote: >> >> Hi, Users >> >> Can i play media while my mod_shout is fetching url stream from the server. >> >> Like, "Please wait we are getting your requested url" >> >> I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ssinyagin at gmail.com Wed Nov 26 16:32:19 2014 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Wed, 26 Nov 2014 14:32:19 +0100 Subject: [Freeswitch-users] Can i play i media while shout is fetching url In-Reply-To: <1417054723570.27045@nexxuspg.com> References: <187E2679-CE2A-40D0-8449-AADBD801F020@jerris.com> <1417027382189.69016@nexxuspg.com> <1417054723570.27045@nexxuspg.com> Message-ID: that's exactly where an external program would help: you start fetching file, and if you see that it's not coming within 2 seconds, you play "please wait". The Go language would be perfect for this, because it has built-in multithreading. On Wed, Nov 26, 2014 at 1:47 PM, Manish Talwar wrote: > Its not like we always want to play "please wait" audio file, but we need to play this file only for those cases where it can take around long time to get a response. > > It will be depend on response time only. > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Stanislav Sinyagin > Sent: 26 November 2014 18:07 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url > > you can make a workaround: with an external script, fetch the whole > audio file, and play the "please wait" while it's getting downloaded, > and then play the audio. Should be quite easy if you use ESL and > multithreading. > > > > > > On Wed, Nov 26, 2014 at 6:12 AM, Manish Talwar > wrote: >> Hi, >> >> We are also looking for same feature to use in our application. We want to play a sound like "please wait..." for those cases where it can take around 10 seconds to get a response. This feature really sound good and must be included in FreeSwitch. >> >> Please let us know whenever this feature will be included in FreeSwitch. >> >> Thanks, >> >> Regards, >> Manish Talwar >> ________________________________________ >> From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Michael Jerris >> Sent: 26 November 2014 03:01 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Can i play i media while shout is fetching url >> >> This sounds like a cool feature, but its not something that you can do right now. I'd be happy to review a patch to add this. >> >>> On Nov 25, 2014, at 2:57 PM, Aqs Younas wrote: >>> >>> Hi, Users >>> >>> Can i play media while my mod_shout is fetching url stream from the server. >>> >>> Like, "Please wait we are getting your requested url" >>> >>> I want to play this file and automatically terminate this file execution when mod_shout starts getting response from requested url . >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Prashanth.Devarajappa at enghouse.com Wed Nov 26 17:01:09 2014 From: Prashanth.Devarajappa at enghouse.com (Prashanth Devarajappa) Date: Wed, 26 Nov 2014 14:01:09 +0000 Subject: [Freeswitch-users] Calleg coming out of conference In-Reply-To: <35badbf7a6e04dfd90841f8b4384f249@UK-MAIL-001.edge.local> References: <35badbf7a6e04dfd90841f8b4384f249@UK-MAIL-001.edge.local> Message-ID: <5c55258e9f434777b0422a6a6f7a4185@UK-MAIL-001.edge.local> Hello, I have a scenario where a call leg/channel has been transferred to a conference and we play a file to this channel using uuid_broadcast. Call leg comes out of conference with message "mod_conference.c:3778 Channel leaving conference, cause: NONE" as soon as play finishes. I don't see anything else in log. It doesn't happen always. ie this is seen around 20% of the time. Any advice will be appreciated. FS Version : 1.2.3 Machine : VM running win2008R2 with 4 cores and 8 GB memory Scenario, Good and bad logs are here https://pastebin.freeswitch.org/23580 Regards Prashanth Devarajappa Senior Software Engineer [cid:image001.png at 01CE94EA.EC69D520] t: +44 118 943 9284 e: PDevarajappa at enghouse.com w: www.enghouseinteractive.com [cid:image002.png at 01CE94EA.EC69D520] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/36854475/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 1045 bytes Desc: image001.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/36854475/attachment-0002.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 5097 bytes Desc: image002.png Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/36854475/attachment-0003.png From ciprian.dosoftei at gmail.com Wed Nov 26 19:13:59 2014 From: ciprian.dosoftei at gmail.com (Ciprian Dosoftei) Date: Wed, 26 Nov 2014 16:13:59 +0000 Subject: [Freeswitch-users] mod_verto behind NAT Message-ID: Hi, I am dealing with a NAT situation related to mod_verto. The switch is hosted in EC2 and the configuration works great with mod_sofia (both for receiving calls from our SIP providers and via webRTC clients). However, the counterpart configuration for mod_verto fails to expose the external IP address in the SDP. Here's the mod_verto configuration we're using: http://pastebin.com/4PDMXvW6 I can confirm the external RTP IP is set properly, however here's the returned SDP: http://pastebin.com/3vzyihRp The verto client is running behind NAT as well, but this doesn't seem to be the root of the problem as Sofia's own implementation of webRTC works fine. It is effectively sending its media to 10.178.x.x rather than the switch's public IP address. Please advise, thank you. -- 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/9aaed94c/attachment.html From msc at freeswitch.org Wed Nov 26 20:52:57 2014 From: msc at freeswitch.org (Michael Collins) Date: Wed, 26 Nov 2014 09:52:57 -0800 Subject: [Freeswitch-users] Calleg coming out of conference In-Reply-To: <5c55258e9f434777b0422a6a6f7a4185@UK-MAIL-001.edge.local> References: <35badbf7a6e04dfd90841f8b4384f249@UK-MAIL-001.edge.local> <5c55258e9f434777b0422a6a6f7a4185@UK-MAIL-001.edge.local> Message-ID: The next step for you is to spin up a test server and use the latest FreeSWITCH and see if you can reproduce this behavior on 1.4.14 (or latest source). Hopefully it just works and you can migrate to 1.4 from FreeSWITCH 1.2 and you are on a pretty old version. -MC On Wed, Nov 26, 2014 at 6:01 AM, Prashanth Devarajappa < Prashanth.Devarajappa at enghouse.com> wrote: > Hello, > > > > I have a scenario where a call leg/channel has been transferred to a > conference and we play a file to this channel using uuid_broadcast. Call > leg comes out of conference with message "mod_conference.c:3778 Channel > leaving conference, cause: NONE" as soon as play finishes. I don?t see > anything else in log. It doesn't happen always. ie this is seen around 20% > of the time. Any advice will be appreciated. > > > > FS Version : 1.2.3 > > Machine : VM running win2008R2 with 4 cores and 8 GB memory > > > > > > Scenario, Good and bad logs are here https://pastebin.freeswitch.org/23580 > > > > > > Regards > > > > Prashanth Devarajappa > Senior Software Engineer > [image: cid:image001.png at 01CE94EA.EC69D520] > t: +44 118 943 9284 > e: PDevarajappa at enghouse.com > w: www.enghouseinteractive.com > [image: cid:image002.png at 01CE94EA.EC69D520] > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/877ca5a7/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 1045 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/877ca5a7/attachment.png -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 5097 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/877ca5a7/attachment-0001.png From dsaraginov at gmail.com Thu Nov 27 00:16:27 2014 From: dsaraginov at gmail.com (Dragan Saraginov) Date: Wed, 26 Nov 2014 22:16:27 +0100 Subject: [Freeswitch-users] Freeswitch and virtual FAX In-Reply-To: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> References: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> Message-ID: Hello Michael, my scenario is for outbound fax and is the following Hylafax > Freeswitch > Telcom . I looked on the internet but I can not find any solution without BRI or Analog cards. Can you please share if you know how can I do this? I am stuck on freetdm, but seems this module works only if I have some BRI or Analog card in the system. On Mon, Nov 24, 2014 at 4:55 PM, Michael Jerris wrote: > assuming you have a provider that supports t.38, sure. > > > On Nov 24, 2014, at 10:46 AM, Dragan Saraginov > wrote: > > > > Hi guys, > > > > I hope someone can help me on the issue that I am facing. I am planing > to implement a virtual FAX machine with freeswitch. My main concern is, > will it be possible to do that without any type of card, analog, digital, > ATA gateway or IAXmodem, but using the t38 support? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141126/c27e72c7/attachment.html From andrea.bartos at virtual-call-center.eu Thu Nov 27 12:40:09 2014 From: andrea.bartos at virtual-call-center.eu (Andrea Bartos) Date: Thu, 27 Nov 2014 10:40:09 +0100 Subject: [Freeswitch-users] dtmf send from mod_verto Message-ID: Hi, Our scenario is the following: A leg: Verto client - FS server B leg: FS server - SIP phone Sending dtmf in the direction from A leg to B leg works, but during the same call, sending dtmf in the opposite direction doesn't. More precisely, on the B leg we have dtmf of type rfc2833. It can be seen from logs, that after receiving dtmf from the SIP phone, the io_routine rtc_send_dtmf() of mod_rtc is called. Our aim is to send dtmf in JSON to Verto client the same way as it works in the direction from A leg to B leg. To solve our issue, we tried to add the io_routine verto_send_dtmf() to mod_verto.c, but it was not called. Could you help us, please? Thanks in advance, Andrea Balog-Bartos -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/0701891a/attachment-0001.html From k4kaleem at gmail.com Thu Nov 27 13:48:13 2014 From: k4kaleem at gmail.com (kaleem rehman) Date: Thu, 27 Nov 2014 10:48:13 +0000 Subject: [Freeswitch-users] how to play a file to recipient In-Reply-To: References: Message-ID: Hi All, I want to play a message to recipient to advise them what line call came in, in Avaya world this is known as Whisper / VDN of Origin Announcement. what I want to achieve is when call hits the Gateway its setup to route call to an extension, when extension answers the call they hear a wav file or a message via (Freeswitch SAY command) saying "customer service" or "Sales" etc... thanks in advance, kaleem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/ba1999ee/attachment.html From matt at inveroak.com Thu Nov 27 13:53:47 2014 From: matt at inveroak.com (Matt Broad) Date: Thu, 27 Nov 2014 10:53:47 +0000 Subject: [Freeswitch-users] how to play a file to recipient In-Reply-To: References: Message-ID: <547702BB.4000805@inveroak.com> Hi, you could use group_confirm_file like below: This will play the file when the call is connected to the bleg. Thanks Matt On 27/11/2014 10:48, kaleem rehman wrote: > Hi All, > > I want to play a message to recipient to advise them what line call > came in, in Avaya world this is known as Whisper / VDN of Origin > Announcement. > > what I want to achieve is when call hits the Gateway its setup to > route call to an extension, when extension answers the call they hear > a wav file or a message via (Freeswitch SAY command) saying "customer > service" or "Sales" etc... > > thanks in advance, > kaleem > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/6499621c/attachment.html From akhilgarg7 at gmail.com Thu Nov 27 14:13:46 2014 From: akhilgarg7 at gmail.com (akhil garg) Date: Thu, 27 Nov 2014 16:43:46 +0530 Subject: [Freeswitch-users] How many concurrent Channels allowed in Freeswitch Message-ID: How Many concurrent channels are allowed in freeswitch? Is there any global parameter or per sip profile parameter in xml files to control this Max Number? Regards, Akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/c611bdff/attachment.html From steveayre at gmail.com Thu Nov 27 14:25:40 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 27 Nov 2014 11:25:40 +0000 Subject: [Freeswitch-users] How many concurrent Channels allowed in Freeswitch In-Reply-To: References: Message-ID: As many as your hardware can handle. FreeSWITCH has adjustable limits in switch.conf.xml, see max-sessions and sessions-per-second https://wiki.freeswitch.org/wiki/XML_Switch_Configuration They can be viewed and modified at runtime with fsctl https://wiki.freeswitch.org/wiki/Mod_commands#fsctl FreeSWITCH may adjust the limits if it finds it can't keep up. On 27 November 2014 at 11:13, akhil garg wrote: > How Many concurrent channels are allowed in freeswitch? > > Is there any global parameter or per sip profile parameter in xml files to > control this Max Number? > > > Regards, > Akhil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/82dad396/attachment.html From telishisheer at gmail.com Thu Nov 27 13:28:36 2014 From: telishisheer at gmail.com (Shisheer Teli) Date: Thu, 27 Nov 2014 15:58:36 +0530 Subject: [Freeswitch-users] Need help for LDAP and Fresswitch integration Message-ID: Dear Team, I configured freeswitch 1.2 stable version successfully and it working fine, but now i want all users should authenticate from LDAP instead of local directory users. I don't know what changes i should do. Please help me its an urgent.... --- Regards, Shisheer T Phone: +91-022 2278 2763 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/7717e5fe/attachment.html From steveayre at gmail.com Thu Nov 27 16:07:35 2014 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 27 Nov 2014 13:07:35 +0000 Subject: [Freeswitch-users] how to play a file to recipient In-Reply-To: References: Message-ID: call say via execute_on_answer perhaps On 27 November 2014 at 10:48, kaleem rehman wrote: > Hi All, > > I want to play a message to recipient to advise them what line call came > in, in Avaya world this is known as Whisper / VDN of Origin Announcement. > > what I want to achieve is when call hits the Gateway its setup to route > call to an extension, when extension answers the call they hear a wav > file or a message via (Freeswitch SAY command) saying "customer service" > or "Sales" etc... > > thanks in advance, > kaleem > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/09f301fa/attachment.html From steveu at coppice.org Thu Nov 27 17:56:38 2014 From: steveu at coppice.org (Steve Underwood) Date: Thu, 27 Nov 2014 22:56:38 +0800 Subject: [Freeswitch-users] Freeswitch and virtual FAX In-Reply-To: References: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> Message-ID: <54773BA6.9050300@coppice.org> How are you trying to connect to your telecom provider? If it is by VoIP, then why are you using freetdm? If its by a PSTN connection you will need freetdm and some type of voice card (ISDN, E1, T1 or analogue) to connect. Do you need to use HylaFAX or is that just one of your options? - You can send and receive FAXes directly by audio or T.38 using FreeSWITCH itself. - You can have HylaFAX use FreeSwitch as its modem bank for FAX, but right now that only works for audio FAX. - If you just want the qeuing scheme that HylaFAX gives you, then there is a package called gofaxip which will let you run a HyfaFAX server, but uses FreeSwitch to actually send and receive the FAXes. Regards, Steve On 11/27/2014 05:16 AM, Dragan Saraginov wrote: > Hello Michael, > > my scenario is for outbound fax and is the following Hylafax > > Freeswitch > Telcom . I looked on the internet but I can not find any > solution without BRI or Analog cards. Can you please share if you know > how can I do this? I am stuck on freetdm, but seems this module works > only if I have some BRI or Analog card in the system. > > On Mon, Nov 24, 2014 at 4:55 PM, Michael Jerris > wrote: > > assuming you have a provider that supports t.38, sure. > > > On Nov 24, 2014, at 10:46 AM, Dragan Saraginov > > wrote: > > > > Hi guys, > > > > I hope someone can help me on the issue that I am facing. I am > planing to implement a virtual FAX machine with freeswitch. My > main concern is, will it be possible to do that without any type > of card, analog, digital, ATA gateway or IAXmodem, but using the > t38 support? > From dsaraginov at gmail.com Thu Nov 27 18:29:02 2014 From: dsaraginov at gmail.com (Dragan Saraginov) Date: Thu, 27 Nov 2014 16:29:02 +0100 Subject: [Freeswitch-users] Freeswitch and virtual FAX In-Reply-To: <54773BA6.9050300@coppice.org> References: <1975F65B-B441-414B-B514-276621824EEF@jerris.com> <54773BA6.9050300@coppice.org> Message-ID: Thanks Steve for the email and your effort to resolve my issue. To clarify things, I want to use HylaFAX with FreeSWITCH. The telco provider is VoIP (SIP) The thing is I can not figure out the outgoing dialplan for this transfer_fax extension. Can you please share if you have any experience with this. Best regards, Dragan On Thu, Nov 27, 2014 at 3:56 PM, Steve Underwood wrote: > How are you trying to connect to your telecom provider? If it is by > VoIP, then why are you using freetdm? If its by a PSTN connection you > will need freetdm and some type of voice card (ISDN, E1, T1 or analogue) > to connect. Do you need to use HylaFAX or is that just one of your options? > > - You can send and receive FAXes directly by audio or T.38 using > FreeSWITCH itself. > - You can have HylaFAX use FreeSwitch as its modem bank for FAX, but > right now that only works for audio FAX. > - If you just want the qeuing scheme that HylaFAX gives you, then there > is a package called gofaxip which will let you run a HyfaFAX server, but > uses FreeSwitch to actually send and receive the FAXes. > > Regards, > Steve > > On 11/27/2014 05:16 AM, Dragan Saraginov wrote: > > Hello Michael, > > > > my scenario is for outbound fax and is the following Hylafax > > > Freeswitch > Telcom . I looked on the internet but I can not find any > > solution without BRI or Analog cards. Can you please share if you know > > how can I do this? I am stuck on freetdm, but seems this module works > > only if I have some BRI or Analog card in the system. > > > > On Mon, Nov 24, 2014 at 4:55 PM, Michael Jerris > > wrote: > > > > assuming you have a provider that supports t.38, sure. > > > > > On Nov 24, 2014, at 10:46 AM, Dragan Saraginov > > > wrote: > > > > > > Hi guys, > > > > > > I hope someone can help me on the issue that I am facing. I am > > planing to implement a virtual FAX machine with freeswitch. My > > main concern is, will it be possible to do that without any type > > of card, analog, digital, ATA gateway or IAXmodem, but using the > > t38 support? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141127/1d0e2b9c/attachment.html From notify.sina at gmail.com Thu Nov 27 22:10:37 2014 From: notify.sina at gmail.com (Notify Me) Date: Thu, 27 Nov 2014 20:10:37 +0100 Subject: [Freeswitch-users] Newbie -- Help Needed Transferring Inbound Caller ID to external SIP Gateway URI Message-ID: Hi! I am very very new to Freeswitch. I am running FreeSWITCH Version 1.5.15b+git~20141120T035109Z~79de78a0fb~64bit (git 79de78a 2014-11-20 03:51:09Z 64bit) on a CentOS 6.6 64-bit Virtual Machine. I am trying to setup freeswitch such that once it gets a call through a sip gateway, it sends the caller ID to another SIP gateway (A URI) to be processed. I am having a lot of difficulty and I am not sure if I am doing the right things correctly, and the logs give so much information it is difficult to see what is happening. Please help me verify if I am correctly configured. I've setup the gateways I expect inbound calls from and freeswitch registers correctly. example: Sofia returns: external::sipgw081 gateway sip:012345081 at sip.sipgwtelecoms.com REGED The gateway I expect to route caller_id_numbers to is defined, but as I was not given a password or asked to register I set it up like so (is this OK?): Sofia returns: external::othersipgw gateway sip:user.name at sip.othersipgw.in NOREG >From what I can see in the documentation I've read, in order to route inbound calls anywhere in freeswitch, Calls that initially come in to the public context and are treated as untrusted?if they are not specifically routed to an extension in the default context, then they are simply disconnected. So I created a file in conf/dailplan/public with the following to dail an extension 1212 in default( I want to route caller IDs with 11 digits, all beginning with number 0): This is where my confusion starts. I both created a file in conf/dialplan/default and edited the conf/dialplan/default.xml to route calls sent to 1212 through to the external gateway URI. I also want the caller_id prefixed with a +234 default.xml: outbound.xml file in conf/dialplan/default: Any assistance gratefully accepted. From krice at freeswitch.org Fri Nov 28 18:03:44 2014 From: krice at freeswitch.org (Ken Rice) Date: Fri, 28 Nov 2014 15:03:44 +0000 Subject: [Freeswitch-users] FreeSWITCH Friday FreeForAll Reminder! Message-ID: <54788ed05774_b924bd318419e5@ip-10-99-170-182.mail> FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888 at conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141128/27e9a289/attachment.html From bordmi at rarus.ru Fri Nov 28 18:41:25 2014 From: bordmi at rarus.ru (=?UTF-8?B?0JHQvtGA0LjRgdC+0LIsINCU0LzQuNGC0YDQuNC5?=) Date: Fri, 28 Nov 2014 19:41:25 +0400 Subject: [Freeswitch-users] High Availability Cluster Module for FreeSWITCH Message-ID: Hi, All! In what state this module is now? -- ? ?????????, ??????? ??????? ????????????? ?? ?????????? VoIP ??????? ????????? ????????????? ?????? ???????? ?????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141128/4af2d6a1/attachment.html From telishisheer at gmail.com Fri Nov 28 12:49:19 2014 From: telishisheer at gmail.com (Shisheer Teli) Date: Fri, 28 Nov 2014 15:19:19 +0530 Subject: [Freeswitch-users] while loading mod_xml_ldap module getting error Message-ID: Dear, while loading mod_xml_ldap module i found following error 2014-11-28 14:55:18.308438 [CRIT] switch_loadable_module.c:1391 Error Loading module /usr/local/freeswitch/mod/mod_xml_ldap.so */usr/local/freeswitch/mod/mod_xml_ldap.so: undefined symbol: lutil_sasl_interact* can help me to troubleshoot -- Regards, Shisheer T -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141128/5e783473/attachment.html From zoell at zoell.us Fri Nov 28 20:06:42 2014 From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=) Date: Fri, 28 Nov 2014 17:06:42 +0000 Subject: [Freeswitch-users] mod_xml_curl sofia domain gateways Message-ID: Hi, What I would like to do is, I need to handle multiple domains. Every domain will have their own gateways and these domains will use only one sip profile. So first when I start FS I got a request: *REQUEST 1:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [d03d1242-7722-11e4-b04c-0f0151b60864] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 17:20:13] Event-Date-GMT = [Fri, 28 Nov 2014 17:20:13 GMT] Event-Date-Timestamp = [1417195213442735] Event-Calling-File = [sofia.c] Event-Calling-Function = [config_sofia] Event-Calling-Line-Number = [3948] Event-Sequence = [18] So this is requesting the sofia config, so I send back that xml: *REPLY 1:*
lot of parameters...
Than it realizes I have a profile specified here, so it requesting something similar, with the profile name key: *REQUEST 2:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213090586] Event-Calling-File = [sofia.c] Event-Calling-Function = [config_sofia] Event-Calling-Line-Number = [3948] Event-Sequence = [934] profile = [MainCCSIP] I am not sure what does it want now but I will send back the same XML, containing my domains (1 for now), the profile settings and no gateways assigned to this profile: *SAME REPLY1:*
lot of parameters...
Please note I excluded all the parameters as there are a lot in the original response. Than I got a new request: *REQUEST 3:* hostname = [dev] section = [configuration] tag_name = [configuration] key_name = [name] key_value = [sofia.conf] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213190626] Event-Calling-File = [sofia.c] Event-Calling-Function = [launch_sofia_worker_thread] Event-Calling-Line-Number = [2658] Event-Sequence = [938] profile = [MainCCSIP] I am wondering why is this request? The only difference is the calling function. Anyway I am sending back the same as above. Then I got an other request: *REQUEST 4:* hostname = [dev] section = [directory] tag_name = [domain] key_name = [name] key_value = [domain1.example.com] Event-Name = [REQUEST_PARAMS] Core-UUID = [199eb63c-7716-11e4-a68f-272cf7c86620] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-28 16:46:53] Event-Date-GMT = [Fri, 28 Nov 2014 16:46:53 GMT] Event-Date-Timestamp = [1417193213330579] Event-Calling-File = [sofia.c] Event-Calling-Function = [launch_sofia_worker_thread] Event-Calling-Line-Number = [2676] Event-Sequence = [939] purpose = [gateways] profile = [MainCCSIP] This seems fine as I have set parse=true in the domain parameters so now it is requesting the gateways for my specific domain1.example.com domain. Now I am sending back the following:
Then I can not see anything in the fs cli related to this gateway. It does not try to connect to it, I can not list it, and also can not see any errors related to this even if I set every trace and debug level to the maximum. I also tried to enclose the gateways node into the domain like: All gateway config here... But I can not see anything which indicates that this is correct/incorrect or what. Is there any problem with this response or am I missing something? I have not send back any other config that FS requests at this stage, just these. Many thanks, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141128/50ab7545/attachment-0001.html From hi-tecc at hotmail.com Sat Nov 29 04:05:59 2014 From: hi-tecc at hotmail.com (DP .) Date: Fri, 28 Nov 2014 20:05:59 -0500 Subject: [Freeswitch-users] Mod LCR SQL sub-query for better routing In-Reply-To: References: <52AA1192.6020005@gmail.com>, <52AB750E.10405@freeswitch.org>, , Message-ID: Victor, I take back my initial response on this old email. We ran into a couple of cases with some carriers and multiple matching prefixes. After actually trying your sub query (modified for Mysql), it actually does return the true lowest rate from a carrier while importantly respecting the longest match per carrier. Unlike the reorder_by_rate function that does not respect the longest match per carrier. That function simply returns the lowest rate, period. So thanks! From: hi-tecc at hotmail.com To: freeswitch-users at lists.freeswitch.org Subject: RE: [Freeswitch-users] Mod LCR SQL sub-query for better routing Date: Mon, 16 Dec 2013 18:36:06 -0500 I agree. This definitely sounds like he simply needed the "reorder_by_rate" param. It will reorder the initial sql results strictly by rate: reorder_by_rate - Forces the LCR module to re-order the query strictly on rate basis. By default this is turned off, but enabling this will always prefer rate over anything else. Beware this may have an adverse effect! I initially had this turned on then quickly realized it would sometimes try to route ALL calls by the lowest rate found. Ex: flowroute lists all calls for the US with a default NPA of "1" at .0098. Now a user trying to call Jamaica with "1876" at a rate of 0.19 (or whatever) will get both flowroute rates returned. The reorder by rate will assume 0.0098 is a valid rate since it will now be the "cheapest" in the list and send the call along its way to flowroute, whom will now bill you at 0.19. Now if you have another carrier in your list with 1876 at 0.15 you can see why this would be a problem. In this case you will always want the longest matched NPANXX rate. Date: Fri, 13 Dec 2013 15:58:54 -0500 From: intralanman at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Mod LCR SQL sub-query for better routing On 12/12/2013 02:42 PM, Victor Chukalovskiy wrote: Hello, For those interested, I added a piece to mod lcr wiki. It makes sorting / routing logic better than default logic: http://wiki.freeswitch.org/wiki/Mod_lcr#Custom_SQL_with_sub-query_-_for_real-life_ratesheet_complexities Why it helps: Rates rates can often be given both on per-NPA or per-NPANXX level depending on the carrier and on the NPA. Moreover, some carriers may have NPANXX rate lower than the corresponding NPA rate, while others will have it inverse. Neither simple ORDER BY rate, prefix; nor ORDER BY prefix, rate; give the truly cheapest route. The LCR logic should be two-step process to accommodate this. Cheers, -Victor Unless I misunderstand what you're saying, this is what the reorder_by_rate param does. You'll always want to pick the longest digit match per carrier. Then you probably want to grab the cheapest overall rate of the matches you got back. -Ray _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141128/01f8ac19/attachment.html From ahabiba at gmail.com Sat Nov 29 03:24:16 2014 From: ahabiba at gmail.com (Ahmed Habiba) Date: Sat, 29 Nov 2014 03:24:16 +0300 Subject: [Freeswitch-users] FreeSWITCH TLS not able to receive calls Message-ID: <069E914C-296C-495C-8A49-54E1A74A866E@gmail.com> Dears, I?ve configured FreeSWITCH with the below version with TLS/SRTPas per the recommendation in page ?https://wiki.freeswitch.org/wiki/SIP_TLS? and it was strait forward, and I was able to connect and make make calls using zoiper, but I was not able to receive any calls after enabling the TLS/SRTP. "FreeSWITCH Version 1.4.13+git~20141103T195300Z~b942d0faa8~64bit (git b942d0f 2014-11-03 19:53:00Z 64bit)? Your kind feedback will be appreciate. Thanks, Ahmed Habiba. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141129/4761e472/attachment.html From zoell at zoell.us Sat Nov 29 21:15:31 2014 From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=) Date: Sat, 29 Nov 2014 18:15:31 +0000 Subject: [Freeswitch-users] mod_xml_curl dialplan not requested Message-ID: Hi, I have a multi tenant multi domain setup. For some reasons my FS won't try to request the dialplan configuration. *xml_curl:* My sofia conf that I send through xml_curl upon request:
My user directory looks like this: *User 1000 at domain 1:*
*User 1000 at domain 2:*
So far the registration for both users in both domains are working and I can register with sip clients. But if I start to dial something, FS won't request for the dialplan. Here is what I can see in the cli trace: freeswitch at dev> tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xd47840 from (udp/192.168.0.132:5060) has 972 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xd47840 (972 bytes) from udp/192.168.0.15:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received INVITE sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) nta.c:3085 agent_recv_request() nta: INVITE (20) going to a default leg nta.c:1348 set_timeout() nta: timer shortened to 200 ms nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:899 nh_create() nua: nh_create: entering nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:280 soa_clone() soa_clone(static::0xb8f480, 0xb7cea0, 0xb85400) called soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called nta.c:4415 nta_leg_tcreate() nta_leg_tcreate(0xdcaf90) soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0xa98310) called soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0xa98310, (nil), 0xb54b3b, 433) called nua_dialog.c:338 nua_dialog_usage_add() nua(0xb85400): adding session usage tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060 tport.c:3492 tport_send_msg() tport_vsend returned 307 nta.c:6789 incoming_reply() nta: sent 100 Trying for INVITE (20) nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: init -> received, received offer soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xa98310, [0x7fe32eca6608], [0x7fe32eca6610], [(nil)]) called nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering 2014-11-29 18:31:52.853532 [NOTICE] switch_channel.c:1055 New Channel sofia/MainCCSIP/1000 at d.2.c.domain.com [fd50511c-77f5-11e4-b25e-c755bd10628e] nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering 2014-11-29 18:31:52.853532 [NOTICE] sofia.c:6946 Hangup sofia/MainCCSIP/ 1000 at d.2.c.domain.com [CS_NEW] [INCOMPATIBLE_DESTINATION] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:879 nua_respond() nua: nua_respond: entering nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called nua_session.c:2318 nua_invite_server_respond() nua: nua_invite_server_respond: entering soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0xa98310) called tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/192.168.0.15:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by name UDP/192.168.0.15:5060 tport.c:3492 tport_send_msg() tport_vsend returned 892 nta.c:6789 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE (20) nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb85400): removing session usage nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state changed: received -> terminated soa.c:356 soa_destroy() soa_destroy(static::0xa98310) called nta.c:4468 nta_leg_destroy() nta_leg_destroy(0xdcaf90) nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_respond tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xaa4690 from (udp/192.168.0.132:5060) has 324 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xaa4690 (324 bytes) from udp/192.168.0.15:5060/sip next=(nil) nta.c:2880 agent_recv_request() nta: received ACK sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) nta.c:3019 agent_recv_request() nta: ACK (20) is going to INVITE (20) 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1633 Session 4 (sofia/MainCCSIP/1000 at d.2.c.domain.com) Ended 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close Channel sofia/MainCCSIP/1000 at d.2.c.domain.com [CS_DESTROY] nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering nta.c:4468 nta_leg_destroy() nta_leg_destroy((nil)) nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_destroy nta.c:1296 agent_timer() nta: timer set next to 1976 ms The only request is coming from time to time is this: hostname = [dev] section = [directory] tag_name = [domain] key_name = [name] key_value = [d.2.c.domain.com] Event-Name = [REQUEST_PARAMS] Core-UUID = [d1af88e4-77f4-11e4-b1f2-c755bd10628e] FreeSWITCH-Hostname = [dev] FreeSWITCH-Switchname = [dev] FreeSWITCH-IPv4 = [192.168.0.132] FreeSWITCH-IPv6 = [::1] Event-Date-Local = [2014-11-29 18:35:05] Event-Date-GMT = [Sat, 29 Nov 2014 18:35:05 GMT] Event-Date-Timestamp = [1417286105312271] Event-Calling-File = [sofia_reg.c] Event-Calling-Function = [sofia_reg_parse_auth] Event-Calling-Line-Number = [2739] Event-Sequence = [898] action = [sip_auth] sip_profile = [MainCCSIP] sip_user_agent = [qutecom/rev-6653cd8f2afd-trunk] sip_auth_username = [1000] sip_auth_realm = [d.2.c.domain.com] sip_auth_nonce = [6ff7849c-77f6-11e4-b27c-c755bd10628e] sip_auth_uri = [sip:d.2.c.domain.com] sip_contact_user = [1000] sip_contact_host = [192.168.0.15] sip_to_user = [1000] sip_to_host = [d.2.c.domain.com] sip_via_protocol = [udp] sip_from_user = [1000] sip_from_host = [d.2.c.domain.com] sip_call_id = [1172541516 at 192.168.0.15] sip_request_host = [d.2.c.domain.com] sip_auth_response = [6836f04faa9f464b8d78f11e88d7ee5d] sip_auth_method = [REGISTER] client_port = [5060] key = [id] user = [1000] domain = [d.2.c.domain.com] ip = [192.168.0.15] Any thoughts why is the dialplan not requested? Thanks, Zoltan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141129/92bda7aa/attachment-0001.html From zoell at zoell.us Sat Nov 29 22:23:48 2014 From: zoell at zoell.us (=?UTF-8?B?Wm9sdMOhbiBTemFiw7M=?=) Date: Sat, 29 Nov 2014 19:23:48 +0000 Subject: [Freeswitch-users] mod_xml_curl dialplan not requested In-Reply-To: References: Message-ID: One addition to this is my modules conf: 2014-11-29 18:15 GMT+00:00 Zolt?n Szab? : > Hi, > > I have a multi tenant multi domain setup. For some reasons my FS won't try > to request the dialplan configuration. > > *xml_curl:* > > > > > > > > > > bindings="configuration"/> > > > > > > > My sofia conf that I send through xml_curl upon request: > >
> > > > > > > > > > > > > > > > alias="false" parse="false"/> > alias="false" parse="false"/> > > > > > > value="_DISABLED_"/> > value="true"/> > value="true"/> > > value="1000"/> > > > > > > > > > > > > > > > > > > value="contact"/> > value="120"/> > value="/usr/local/freeswitch/script/emailvm.py"/> > > > > > > value="/usr/local/freeswitch/scripts/emailvm.py"/> > > > > > > data="inbound_call=true" direction="inbound"/> > data="outbound_call=true" direction="outbound"/> > data="gw=gwname"/> > > > > > > > > > > > > > > > >
> > > > My user directory looks like this: > > *User 1000 at domain 1:* > >
> > > > > > > > > > > > > > > > > > value='123456'/> > value='123456'/> > name="vm-keep-local-after-email" value="true"/> > value="mp3"/> > > > name="outbound_caller_id_name" value="Unknown"/> > name="outbound_caller_id_number" value="0000000000"/> > name="internal_caller_id_name" value="Unknown"/> > value="true"/> > value="domestic,international,local"/> > value="1000"/> > value='nat-connectile-dysfunction'/> > value="false"/> > name="sip-allow-multiple-registrations" value="false"/> > > value="something"/> > value="1000 at d.1.c.domain.com"/> > name="transfer_fallback_extension" value="operator"/> > > > > > > >
> > > > *User 1000 at domain 2:* > >
> > > > > > > > > > > > > > > > > > value='654321'/> > value='123456'/> > name="vm-keep-local-after-email" value="true"/> > value="mp3"/> > > > name="outbound_caller_id_name" value="Unknown"/> > name="outbound_caller_id_number" value="0000000000"/> > name="internal_caller_id_name" value="Unknown"/> > value="true"/> > value="domestic,international,local"/> > value="0"/> > value='nat-connectile-dysfunction'/> > value="false"/> > name="sip-allow-multiple-registrations" value="false"/> > > value="something"/> > value="1000 at d.2.c.domain.com"/> > name="transfer_fallback_extension" value="operator"/> > > > > > > >
> > > > So far the registration for both users in both domains are working and I > can register with sip clients. But if I start to dial something, FS won't > request for the dialplan. > > Here is what I can see in the cli trace: > freeswitch at dev> tport.c:2749 tport_wakeup_pri() > tport_wakeup_pri(0xb92ee0): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xd47840 > from (udp/192.168.0.132:5060) has 972 bytes, veclen = 1 > tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xd47840 (972 > bytes) from udp/192.168.0.15:5060/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received INVITE > sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) > nta.c:3085 agent_recv_request() nta: INVITE (20) going to a default leg > nta.c:1348 set_timeout() nta: timer shortened to 200 ms > nua_server.c:102 nua_stack_process_request() nua: > nua_stack_process_request: entering > nua_stack.c:899 nh_create() nua: nh_create: entering > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:280 soa_clone() soa_clone(static::0xb8f480, 0xb7cea0, 0xb85400) > called > soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called > nta.c:4415 nta_leg_tcreate() nta_leg_tcreate(0xdcaf90) > soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0xa98310) > called > soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0xa98310, > (nil), 0xb54b3b, 433) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0xb85400): adding session usage > tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/ > 192.168.0.15:5060 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 > tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by > name UDP/192.168.0.15:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 307 > nta.c:6789 incoming_reply() nta: sent 100 Trying for INVITE (20) > nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state > changed: init -> received, received offer > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xa98310, > [0x7fe32eca6608], [0x7fe32eca6610], [(nil)]) called > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > 2014-11-29 18:31:52.853532 [NOTICE] switch_channel.c:1055 New Channel > sofia/MainCCSIP/1000 at d.2.c.domain.com > [fd50511c-77f5-11e4-b25e-c755bd10628e] > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2014-11-29 18:31:52.853532 [NOTICE] sofia.c:6946 Hangup sofia/MainCCSIP/ > 1000 at d.2.c.domain.com [CS_NEW] [INCOMPATIBLE_DESTINATION] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:879 nua_respond() nua: nua_respond: entering > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called > nua_session.c:2318 nua_invite_server_respond() nua: > nua_invite_server_respond: entering > soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0xa98310) > called > tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/ > 192.168.0.15:5060 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 > tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by > name UDP/192.168.0.15:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 892 > nta.c:6789 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE > (20) > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb85400): removing > session usage > nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state > changed: received -> terminated > soa.c:356 soa_destroy() soa_destroy(static::0xa98310) called > nta.c:4468 nta_leg_destroy() nta_leg_destroy(0xdcaf90) > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_respond > tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xaa4690 > from (udp/192.168.0.132:5060) has 324 bytes, veclen = 1 > tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xaa4690 (324 > bytes) from udp/192.168.0.15:5060/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received ACK > sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) > nta.c:3019 agent_recv_request() nta: ACK (20) is going to INVITE (20) > 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1633 Session 4 > (sofia/MainCCSIP/1000 at d.2.c.domain.com) Ended > 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close > Channel sofia/MainCCSIP/1000 at d.2.c.domain.com [CS_DESTROY] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > nta.c:4468 nta_leg_destroy() nta_leg_destroy((nil)) > nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_destroy > nta.c:1296 agent_timer() nta: timer set next to 1976 ms > > > The only request is coming from time to time is this: > hostname = [dev] > section = [directory] > tag_name = [domain] > key_name = [name] > key_value = [d.2.c.domain.com] > Event-Name = [REQUEST_PARAMS] > Core-UUID = [d1af88e4-77f4-11e4-b1f2-c755bd10628e] > FreeSWITCH-Hostname = [dev] > FreeSWITCH-Switchname = [dev] > FreeSWITCH-IPv4 = [192.168.0.132] > FreeSWITCH-IPv6 = [::1] > Event-Date-Local = [2014-11-29 18:35:05] > Event-Date-GMT = [Sat, 29 Nov 2014 18:35:05 GMT] > Event-Date-Timestamp = [1417286105312271] > Event-Calling-File = [sofia_reg.c] > Event-Calling-Function = [sofia_reg_parse_auth] > Event-Calling-Line-Number = [2739] > Event-Sequence = [898] > action = [sip_auth] > sip_profile = [MainCCSIP] > sip_user_agent = [qutecom/rev-6653cd8f2afd-trunk] > sip_auth_username = [1000] > sip_auth_realm = [d.2.c.domain.com] > sip_auth_nonce = [6ff7849c-77f6-11e4-b27c-c755bd10628e] > sip_auth_uri = [sip:d.2.c.domain.com] > sip_contact_user = [1000] > sip_contact_host = [192.168.0.15] > sip_to_user = [1000] > sip_to_host = [d.2.c.domain.com] > sip_via_protocol = [udp] > sip_from_user = [1000] > sip_from_host = [d.2.c.domain.com] > sip_call_id = [1172541516 at 192.168.0.15] > sip_request_host = [d.2.c.domain.com] > sip_auth_response = [6836f04faa9f464b8d78f11e88d7ee5d] > sip_auth_method = [REGISTER] > client_port = [5060] > key = [id] > user = [1000] > domain = [d.2.c.domain.com] > ip = [192.168.0.15] > > Any thoughts why is the dialplan not requested? > > Thanks, > Zoltan > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141129/e7fa5fc3/attachment-0001.html From idokan at gmail.com Sun Nov 30 14:25:08 2014 From: idokan at gmail.com (ik) Date: Sun, 30 Nov 2014 13:25:08 +0200 Subject: [Freeswitch-users] multi tenet user directory management best practice Message-ID: Hello, This question raises a lot, but no simple answer, so I'm trying to ask it differently. Let's say I have 5 FS servers and Opensips/Kamailio as a load balancer. I'm afraid that creating it using xml files is not dynamic enough, and requires duplications. mod_xml_rpc does not store that information on each update of the server itself, and there is no API for providing database management that I could find. So what is the best practice regarding of creating a user directory that all of the PBX will be able to handle ? Thanks Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20141130/f9ea941e/attachment.html From covici at ccs.covici.com Sun Nov 30 14:43:01 2014 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sun, 30 Nov 2014 06:43:01 -0500 Subject: [Freeswitch-users] echo cancellation and mod_conference Message-ID: <21728.1417347781@ccs.covici.com> Hi. I have some sip callers who call in on what are apparently bad lines and get echo when calling into conferences -- not freeswitch ones yet. So, I am wondering in mod_conference is there any attempt to do echo cancellation, if so I will try to have them call into an fs one instead. Its intermittent, so I wanted to find out first before I test. 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 covici at ccs.covici.com From steveayre at gmail.com Sun Nov 30 15:43:05 2014 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 30 Nov 2014 12:43:05 +0000 Subject: [Freeswitch-users] mod_xml_curl dialplan not requested In-Reply-To: References: Message-ID: Your call is being rejected with 488 Not Acceptable Here. That means a codec error - the client isn't offerring any of the codecs you've configured FreeSWITCH to accept. Such calls get rejected immediately, before hitting the dialplan (unless using late-negotiation). Incidentally, don't use the sofia stack logging (sofia loglevel). Use the FreeSWITCH logging ("/log debug" in fs_cli). The sofia stack is rather verbose and unreadable, and doesn't tell you anything about what FreeSWITCH is doing at the higher level. Looking inside the stack is something to do if the FreeSWITCH log isn't showing you the error. The FreeSWITCH log'll show the SDP and the codec negotiation, so you'll see the error there. On 29 November 2014 at 18:15, Zolt?n Szab? wrote: > Hi, > > I have a multi tenant multi domain setup. For some reasons my FS won't try > to request the dialplan configuration. > > *xml_curl:* > > > > > > > > > > bindings="configuration"/> > > > > > > > My sofia conf that I send through xml_curl upon request: > >
> > > > > > > > > > > > > > > > alias="false" parse="false"/> > alias="false" parse="false"/> > > > > > > value="_DISABLED_"/> > value="true"/> > value="true"/> > > value="1000"/> > > > > > > > > > > > > > > > > > > value="contact"/> > value="120"/> > value="/usr/local/freeswitch/script/emailvm.py"/> > > > > > > value="/usr/local/freeswitch/scripts/emailvm.py"/> > > > > > > data="inbound_call=true" direction="inbound"/> > data="outbound_call=true" direction="outbound"/> > data="gw=gwname"/> > > > > > > > > > > > > > > > >
> > > > My user directory looks like this: > > *User 1000 at domain 1:* > >
> > > > > > > > > > > > > > > > > > value='123456'/> > value='123456'/> > name="vm-keep-local-after-email" value="true"/> > value="mp3"/> > > > name="outbound_caller_id_name" value="Unknown"/> > name="outbound_caller_id_number" value="0000000000"/> > name="internal_caller_id_name" value="Unknown"/> > value="true"/> > value="domestic,international,local"/> > value="1000"/> > value='nat-connectile-dysfunction'/> > value="false"/> > name="sip-allow-multiple-registrations" value="false"/> > > value="something"/> > value="1000 at d.1.c.domain.com"/> > name="transfer_fallback_extension" value="operator"/> > > > > > > >
> > > > *User 1000 at domain 2:* > >
> > > > > > > > > > > > > > > > > > value='654321'/> > value='123456'/> > name="vm-keep-local-after-email" value="true"/> > value="mp3"/> > > > name="outbound_caller_id_name" value="Unknown"/> > name="outbound_caller_id_number" value="0000000000"/> > name="internal_caller_id_name" value="Unknown"/> > value="true"/> > value="domestic,international,local"/> > value="0"/> > value='nat-connectile-dysfunction'/> > value="false"/> > name="sip-allow-multiple-registrations" value="false"/> > > value="something"/> > value="1000 at d.2.c.domain.com"/> > name="transfer_fallback_extension" value="operator"/> > > > > > > >
> > > > So far the registration for both users in both domains are working and I > can register with sip clients. But if I start to dial something, FS won't > request for the dialplan. > > Here is what I can see in the cli trace: > freeswitch at dev> tport.c:2749 tport_wakeup_pri() > tport_wakeup_pri(0xb92ee0): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xd47840 > from (udp/192.168.0.132:5060) has 972 bytes, veclen = 1 > tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xd47840 (972 > bytes) from udp/192.168.0.15:5060/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received INVITE > sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) > nta.c:3085 agent_recv_request() nta: INVITE (20) going to a default leg > nta.c:1348 set_timeout() nta: timer shortened to 200 ms > nua_server.c:102 nua_stack_process_request() nua: > nua_stack_process_request: entering > nua_stack.c:899 nh_create() nua: nh_create: entering > nua_common.c:108 nh_create_handle() nua: nh_create_handle: entering > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:280 soa_clone() soa_clone(static::0xb8f480, 0xb7cea0, 0xb85400) > called > soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called > nta.c:4415 nta_leg_tcreate() nta_leg_tcreate(0xdcaf90) > soa.c:1302 soa_init_offer_answer() soa_init_offer_answer(static::0xa98310) > called > soa.c:1171 soa_set_remote_sdp() soa_set_remote_sdp(static::0xa98310, > (nil), 0xb54b3b, 433) called > nua_dialog.c:338 nua_dialog_usage_add() nua(0xb85400): adding session usage > tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/ > 192.168.0.15:5060 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 > tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by > name UDP/192.168.0.15:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 307 > nta.c:6789 incoming_reply() nta: sent 100 Trying for INVITE (20) > nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state > changed: init -> received, received offer > soa.c:1098 soa_get_remote_sdp() soa_get_remote_sdp(static::0xa98310, > [0x7fe32eca6608], [0x7fe32eca6610], [(nil)]) called > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > 2014-11-29 18:31:52.853532 [NOTICE] switch_channel.c:1055 New Channel > sofia/MainCCSIP/1000 at d.2.c.domain.com > [fd50511c-77f5-11e4-b25e-c755bd10628e] > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > 2014-11-29 18:31:52.853532 [NOTICE] sofia.c:6946 Hangup sofia/MainCCSIP/ > 1000 at d.2.c.domain.com [CS_NEW] [INCOMPATIBLE_DESTINATION] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:879 nua_respond() nua: nua_respond: entering > nua_params.c:480 nua_stack_set_params() nua: nua_stack_set_params: entering > soa.c:403 soa_set_params() soa_set_params(static::0xa98310, ...) called > nua_session.c:2318 nua_invite_server_respond() nua: > nua_invite_server_respond: entering > soa.c:1214 soa_clear_remote_sdp() soa_clear_remote_sdp(static::0xa98310) > called > tport.c:3257 tport_tsend() tport_tsend(0xb92ee0) tpn = UDP/ > 192.168.0.15:5060 > tport.c:4046 tport_resolve() tport_resolve addrinfo = 192.168.0.15:5060 > tport.c:4680 tport_by_addrinfo() tport_by_addrinfo(0xb92ee0): not found by > name UDP/192.168.0.15:5060 > tport.c:3492 tport_send_msg() tport_vsend returned 892 > nta.c:6789 incoming_reply() nta: sent 488 Not Acceptable Here for INVITE > (20) > nua_dialog.c:397 nua_dialog_usage_remove_at() nua(0xb85400): removing > session usage > nua_session.c:4137 signal_call_state_change() nua(0xb85400): call state > changed: received -> terminated > soa.c:356 soa_destroy() soa_destroy(static::0xa98310) called > nta.c:4468 nta_leg_destroy() nta_leg_destroy(0xdcaf90) > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:359 nua_application_event() nua: nua_application_event: > entering > nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_respond > tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0xb92ee0): events IN > tport.c:2864 tport_recv_event() tport_recv_event(0xb92ee0) > tport.c:3205 tport_recv_iovec() tport_recv_iovec(0xb92ee0) msg 0xaa4690 > from (udp/192.168.0.132:5060) has 324 bytes, veclen = 1 > tport.c:3023 tport_deliver() tport_deliver(0xb92ee0): msg 0xaa4690 (324 > bytes) from udp/192.168.0.15:5060/sip next=(nil) > nta.c:2880 agent_recv_request() nta: received ACK > sip:123456789 at d.2.c.domain.com SIP/2.0 (CSeq 20) > nta.c:3019 agent_recv_request() nta: ACK (20) is going to INVITE (20) > 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1633 Session 4 > (sofia/MainCCSIP/1000 at d.2.c.domain.com) Ended > 2014-11-29 18:31:52.853532 [NOTICE] switch_core_session.c:1637 Close > Channel sofia/MainCCSIP/1000 at d.2.c.domain.com [CS_DESTROY] > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:342 nua_handle_bind() nua: nua_handle_bind: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:366 nua_handle_magic() nua: nua_handle_magic: entering > nua.c:921 nua_handle_destroy() nua: nua_handle_destroy: entering > nta.c:4468 nta_leg_destroy() nta_leg_destroy((nil)) > nua_stack.c:529 nua_signal() nua(0xb85400): sent signal r_destroy > nta.c:1296 agent_timer() nta: timer set next to 1976 ms > > > The only request is coming from time to time is this: > hostname = [dev] > section = [directory] > tag_name = [domain] > key_name = [name] > key_value = [d.2.c.domain.com] > Event-Name = [REQUEST_PARAMS] > Core-UUID = [d1af88e4-77f4-11e4-b1f2-c755bd10628e] > FreeSWITCH-Hostname = [dev] > FreeSWITCH-Switchname = [dev] > FreeSWITCH-IPv4 = [192.168.0.132] > FreeSWITCH-IPv6 = [::1] > Event-Date-Local = [2014-11-29 18:35:05] > Event-Date-GMT = [Sat, 29 Nov 2014 18:35:05 GMT] > Event-Date-Timestamp = [1417286105312271] > Event-Calling-File = [sofia_reg.c] > Event-Calling-Function = [sofia_reg_parse_auth] > Event-Calling-Line-Number = [2739] > Event-Sequence = [898] > action = [sip_auth] > sip_profile = [MainCCSIP] > sip_user_agent = [qutecom/rev-6653cd8f2afd-trunk] > sip_auth_username = [1000] > sip_auth_realm = [d.2.c.domain.com] > sip_auth_nonce = [6ff7849c-77f6-11e4-b27c-c755bd10628e] > sip_auth_uri = [sip:d.2.c.domain.com] > sip_contact_user = [1000] > sip_contact_host = [192.168.0.15] > sip_to_user = [1000] > sip_to_host = [d.2.c.domain.com] > sip_via_protocol = [udp] > sip_from_user = [1000] > sip_from_host = [d.2.c.domain.com] > sip_call_id = [1172541516 at 192.168.0.15] > sip_request_host = [d.2.c.domain.com] > sip_auth_response = [6836f04faa9f464b8d78f11e88d7ee5d] > sip_auth_method = [REGISTER] > client_port = [5060] > key = [id] > user = [1000] > domain = [d.2.c.domain.com] > ip = [192.168.0.15] > > Any thoughts why is the dialplan not requested? > > Thanks, > Zoltan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... 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