From ira at connectmevoice.com Wed May 1 01:12:57 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Tue, 30 Apr 2013 17:12:57 -0400 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: References: <20130427161037.f5b5b1f7@mail.tritonwest.net> Message-ID: When I am building the esl.dll and ManagedEsl.dll, first I load the Freeswitch 2010 solution, select the build configuration of Release, x64 and build the solution. Then I load the ManagedEsl solution and so the same. Is this correct? Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Mon, Apr 29, 2013 at 11:29 AM, Gregor Nanger wrote: > Great Dave! > > Thank you. > > I also had hard time compiling DLLs and also have memory corruption.. Do > not remember exactly, but it has to be right combination of build > configuration... > > Gregor > > > 2013/4/27 Dave R. Kompel > >> ** >> Same thing happened to me. It seems that for some reason if I build it on >> the same machine, and from the same clone that I build the rest of FS I get >> the same results. So every time there is a major change to the ESL stuff, I >> build from fresh clone on a different machine, where nothing else in the >> tree has been built, verfy size of the native DLL, zip, and push up there, >> since I use it in a number of projects. >> >> If I can ever figure out the NuGet powershell magic to build a script >> into a NuGet package that will add the build step to also copy the native >> interop dll as well as the managed one, I'll push a HuGet package up with >> it, and have a CI build trigger on changes to the ESL library. >> >> --Dave >> >> ------------------------------ >> *From:* Ira Tessler [mailto:ira at connectmevoice.com] >> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org >> ] >> *Sent:* Sat, 27 Apr 2013 05:32:41 -0700 >> >> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >> (Windows) >> >> Ok Thanks for the binaries. They work great! I complied them myself. I >> set the build configuration to release and x64. When my app tries to access >> the esl.dll, I get a memory corruption (i don't remember the exact >> exception) exception. It drives me crazy! :) >> >> Ira Tessler >> Lead Software Engineer >> ConnectMe >> (732) 490-9007 x2 >> ira at connectmevoice.com >> >> >> On Fri, Apr 26, 2013 at 12:44 PM, Dave R. Kompel wrote: >> >>> ** >>> Ira, >>> >>> It is a pain, since you have to have our dev box set up right. For that >>> reason I keep this: http://download.drknetworking.com/eslmanaged.zip around >>> for ppl, it has both native wrappers in it. >>> >>> --Dave >>> >>> ------------------------------ >>> *From:* Ira Tessler [mailto:ira at connectmevoice.com] >>> *To:* FreeSWITCH Users Help [mailto: >>> freeswitch-users at lists.freeswitch.org] >>> *Sent:* Fri, 26 Apr 2013 04:18:22 -0700 >>> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >>> (Windows) >>> >>> I am having the same trouble building the ESL. Would you be able to post >>> the steps you did to build both the 32 and 64 bit versions is esl.dll and >>> managedesl.dll? >>> >>> Thanks for the binaries! >>> >>> --Ira >>> >>> Ira Tessler >>> Lead Software Engineer >>> ConnectMe >>> (732) 490-9007 x2 >>> ira at connectmevoice.com >>> >>> >>> On Mon, Feb 4, 2013 at 8:25 PM, Dave R. Kompel wrote: >>> >>>> ** >>>> Getting it to swig right, and getting the native DLL built for X64 was >>>> give me a problem, till I did it on a diferent box. >>>> >>>> I notice you added the VS2012 projects for it as well about an hour ago >>>> :) Thanks. >>>> >>>> --Dave >>>> >>>> ------------------------------ >>>> *From:* Jeff Lenk [mailto:jeff at jefflenk.com] >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Sent:* Mon, 04 Feb 2013 16:37:20 -0800 >>>> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >>>> (Windows) >>>> >>>> Dave I'm curious what was wrong? The solution should build without >>>> changing >>>> anything. >>>> >>>> You have to build FreeSWITCH for the platform you want then open the esl >>>> solution and select the platform you want. The only caveat here is that >>>> because you are building code that is platform specific that you have to >>>> specify the correct platform and not "Any Cpu" or "Mixed Platform". >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.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://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 >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130430/96021026/attachment.html From drk at drkngs.net Wed May 1 03:00:19 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 30 Apr 2013 16:00:19 -0700 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: Message-ID: <20130430230019.60e244cc@mail.tritonwest.net> I don't remember each time I have to figure it out. However the one thing that I do remember is that you can only build it from a clean clone, if you have built FS there first, the build will fail, cause output files conflict. --Dave _____ From: Ira Tessler [mailto:ira at connectmevoice.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 30 Apr 2013 14:12:57 -0700 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) When I am building the esl.dll and ManagedEsl.dll, first I load the Freeswitch 2010 solution, select the build configuration of Release, x64 and build the solution. Then I load the ManagedEsl solution and so the same. Is this correct? Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Mon, Apr 29, 2013 at 11:29 AM, Gregor Nanger wrote: Great Dave! Thank you. I also had hard time compiling DLLs and also have memory corruption.. Do not remember exactly, but it has to be right combination of build configuration... Gregor 2013/4/27 Dave R. Kompel Same thing happened to me. It seems that for some reason if I build it on the same machine, and from the same clone that I build the rest of FS I get the same results. So every time there is a major change to the ESL stuff, I build from fresh clone on a different machine, where nothing else in the tree has been built, verfy size of the native DLL, zip, and push up there, since I use it in a number of projects. If I can ever figure out the NuGet powershell magic to build a script into a NuGet package that will add the build step to also copy the native interop dll as well as the managed one, I'll push a HuGet package up with it, and have a CI build trigger on changes to the ESL library. --Dave _____ From: Ira Tessler [mailto:ira at connectmevoice.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 27 Apr 2013 05:32:41 -0700 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) Ok Thanks for the binaries. They work great! I complied them myself. I set the build configuration to release and x64. When my app tries to access the esl.dll, I get a memory corruption (i don't remember the exact exception) exception. It drives me crazy! :) Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Fri, Apr 26, 2013 at 12:44 PM, Dave R. Kompel wrote: Ira, It is a pain, since you have to have our dev box set up right. For that reason I keep this: http://download.drknetworking.com/eslmanaged.zip around for ppl, it has both native wrappers in it. --Dave _____ From: Ira Tessler [mailto:ira at connectmevoice.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 26 Apr 2013 04:18:22 -0700 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) I am having the same trouble building the ESL. Would you be able to post the steps you did to build both the 32 and 64 bit versions is esl.dll and managedesl.dll? Thanks for the binaries! --Ira Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Mon, Feb 4, 2013 at 8:25 PM, Dave R. Kompel wrote: Getting it to swig right, and getting the native DLL built for X64 was give me a problem, till I did it on a diferent box. I notice you added the VS2012 projects for it as well about an hour ago :) Thanks. --Dave _____ From: Jeff Lenk [mailto:jeff at jefflenk.com] To: freeswitch-users at lists.freeswitch.org Sent: Mon, 04 Feb 2013 16:37:20 -0800 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) Dave I'm curious what was wrong? The solution should build without changing anything. You have to build FreeSWITCH for the platform you want then open the esl solution and select the platform you want. The only caveat here is that because you are building code that is platform specific that you have to specify the correct platform and not "Any Cpu" or "Mixed Platform". -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.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://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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130430/3a221626/attachment-0001.html From lists at kavun.ch Wed May 1 04:04:42 2013 From: lists at kavun.ch (Emrah) Date: Tue, 30 Apr 2013 17:04:42 -0700 Subject: [Freeswitch-users] Bind to multiple ports with Sofia In-Reply-To: References: Message-ID: <6678CCF7-CD89-4B9B-8A19-3951F8975BB6@kavun.ch> Hi Ken, Even with the force-domain option, I can register on a different port but will not be found via sofia_contact user at domain. My impression is that only one profile can alter the register table. If the 5070 profile is up first, than my registrations will work on port 5070 only. If it's the main one on 5060 that comes up first, my contacts on 5070 disappear. Any advice would be very much appreciated. Thanks a bunch for your help, Emrah On Apr 28, 2013, at 7:14 AM, Ken Rice wrote: > You can have any number of domains on different profiles... > > If you just need 1 domain everywhere check the force domain options on the > sip profile configs (I believe they are used in the default example configs > > > On 4/28/13 8:42 AM, "Emrah" wrote: > >> Hi guys, >> >> I need to bind to multiple ports with Sofia. If I start one profile per port, >> I cannot get a consistent registration table working. A domain can only be >> aliased to one profile if my understanding is correct. >> >> If you have a working set up, I'd appreciate some advice. >> >> Thanks, >> Emrah >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net #freeswitch > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From krice at freeswitch.org Wed May 1 05:31:16 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 30 Apr 2013 20:31:16 -0500 Subject: [Freeswitch-users] Bind to multiple ports with Sofia In-Reply-To: <6678CCF7-CD89-4B9B-8A19-3951F8975BB6@kavun.ch> References: <6678CCF7-CD89-4B9B-8A19-3951F8975BB6@kavun.ch> Message-ID: <3096D814-BFB0-4D7E-8320-BC676C60C21B@freeswitch.org> theres a way to do it, i think it might be */user at domain i forget tho.... Ken Sent from my iPad On Apr 30, 2013, at 19:04, Emrah wrote: > Hi Ken, > > Even with the force-domain option, I can register on a different port but will not be found via sofia_contact user at domain. > My impression is that only one profile can alter the register table. If the 5070 profile is up first, than my registrations will work on port 5070 only. If it's the main one on 5060 that comes up first, my contacts on 5070 disappear. > > Any advice would be very much appreciated. > > Thanks a bunch for your help, > Emrah > On Apr 28, 2013, at 7:14 AM, Ken Rice wrote: > >> You can have any number of domains on different profiles... >> >> If you just need 1 domain everywhere check the force domain options on the >> sip profile configs (I believe they are used in the default example configs >> >> >> On 4/28/13 8:42 AM, "Emrah" wrote: >> >>> Hi guys, >>> >>> I need to bind to multiple ports with Sofia. If I start one profile per port, >>> I cannot get a consistent registration table working. A domain can only be >>> aliased to one profile if my understanding is correct. >>> >>> If you have a working set up, I'd appreciate some advice. >>> >>> Thanks, >>> Emrah >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From msc at freeswitch.org Wed May 1 06:15:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 30 Apr 2013 19:15:38 -0700 Subject: [Freeswitch-users] Bind to multiple ports with Sofia In-Reply-To: <3096D814-BFB0-4D7E-8320-BC676C60C21B@freeswitch.org> References: <6678CCF7-CD89-4B9B-8A19-3951F8975BB6@kavun.ch> <3096D814-BFB0-4D7E-8320-BC676C60C21B@freeswitch.org> Message-ID: User at domain/* See also conf/directory/default.xml where dial-string param is defined. -MC On Tuesday, April 30, 2013, Ken Rice wrote: > theres a way to do it, i think it might be */user at domain i forget tho.... > > Ken > Sent from my iPad > > On Apr 30, 2013, at 19:04, Emrah wrote: > >> Hi Ken, >> >> Even with the force-domain option, I can register on a different port but will not be found via sofia_contact user at domain. >> My impression is that only one profile can alter the register table. If the 5070 profile is up first, than my registrations will work on port 5070 only. If it's the main one on 5060 that comes up first, my contacts on 5070 disappear. >> >> Any advice would be very much appreciated. >> >> Thanks a bunch for your help, >> Emrah >> On Apr 28, 2013, at 7:14 AM, Ken Rice wrote: >> >>> You can have any number of domains on different profiles... >>> >>> If you just need 1 domain everywhere check the force domain options on the >>> sip profile configs (I believe they are used in the default example configs >>> >>> >>> On 4/28/13 8:42 AM, "Emrah" wrote: >>> >>>> Hi guys, >>>> >>>> I need to bind to multiple ports with Sofia. If I start one profile per port, >>>> I cannot get a consistent registration table working. A domain can only be >>>> aliased to one profile if my understanding is correct. >>>> >>>> If you have a working set up, I'd appreciate some advice. >>>> >>>> Thanks, >>>> Emrah >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> irc.freenode.net #freeswitch >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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< -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130430/6e4106d3/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed May 1 06:28:24 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 1 May 2013 03:28:24 +0100 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: References: Message-ID: I got 99 patents but SIP ain't one. Cal On Tue, Apr 30, 2013 at 2:06 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > Thought I'd share this with you all... > > So there I was, quietly enjoying myself on theregister, when I stumbled > upon what can only be described, as a complete and utter fail. > > > http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ > > Responses so far? > > <@SwK> lololol > * foxx2 facepalm > BT come up with a regular stream of patents with no basis in > reality > they shouldn't be able to patent a standard protocol > > I'm fairly sure my immediate response would be to print out 50 copies of > [1] and send it via recorded delivery. > > It would be interesting to hear from anyone who has already been contacted > by BT about this, and how they plan to deal with it. > > I've had enough of the internet for one day ;/ > > Cal > > [1] > http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfwZk1r8alg0o1_500.png > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/7834c9fc/attachment.html From sirimmfs at gmail.com Wed May 1 06:41:01 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 1 May 2013 12:41:01 +1000 Subject: [Freeswitch-users] User settings in directory for unauthenticated users Message-ID: Hi, I have a system where I use Open ACL, and allow users to register without any passwords required (one reason being all are trusted devices withing the closed network) However, I would like to restrict outgoing calls, and was hoping to use the 'toll_allow' variable, and have set the same for the user in the directory. However, in the dialplan, I see that toll_allow is not set for the extension - when I experimented by removing the open acl and forcing authentication, toll_allow is set. My question is, if this is expected behavior, is there any other way I could pass on user specific settings to the dialplan in an OpenACL environment? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/239b54c0/attachment-0001.html From michel.brabants at gmail.com Wed May 1 13:26:26 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Wed, 1 May 2013 11:26:26 +0200 Subject: [Freeswitch-users] User settings in directory for unauthenticated users In-Reply-To: References: Message-ID: It works for me if I remember correctly. You should have the cidr-attribute set on the user and the ip is then used to match it with the user. Michel Op 1-mei-2013 04:43 schreef "Siri MM" het volgende: > Hi, > > I have a system where I use Open ACL, and allow users to register > without any passwords required (one reason being all are trusted devices > withing the closed network) > > However, I would like to restrict outgoing calls, and was hoping to use > the 'toll_allow' variable, and have set the same for the user in the > directory. However, in the dialplan, I see that toll_allow is not set for > the extension - when I experimented by removing the open acl and forcing > authentication, toll_allow is set. > > My question is, if this is expected behavior, is there any other way I > could pass on user specific settings to the dialplan in an OpenACL > environment? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130501/59c75b2a/attachment.html From orn at arnarson.net Wed May 1 16:01:22 2013 From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=) Date: Wed, 1 May 2013 12:01:22 +0000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: Message-ID: Thank you very much. An oversight that cost hours of work! Instant change in behavior. Regards, ?rn On Tue, Apr 30, 2013 at 3:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The variable is sip_execute_on_image > > > > On Tue, Apr 30, 2013 at 5:35 AM, ?rn Arnarson wrote: > >> Hello, >> >> I'm trying to get T38 gatewaying to work. I have a call coming in with >> FAX audio, and an endpoint that needs T38. The endpoint will send a T38 >> re-invite, to which FS responds 100 trying, and after 40 seconds the >> endpoint will give up (citing stale 100 trying). >> >> I set these variables before briding: >> >> >> >> And then I bridge like this (dialstring is sofia/profile/number at ip:port): >> >> >> I've tried all variants I can think of, and always have the same result, >> except for when fax_enable_t38 = false, in which case FS rejects the >> re-invite. >> >> I found some old discussions on the mailing list alluding to the same >> problem, but not much in the way of solutions. >> >> Can anybody give me a pointer as to what I'm doing wrong? I was running a >> git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the same >> results. >> >> Here's what happens in FS. At the very end you can see the Re-INVITE, and >> then nothing happens post that point. >> http://pastebin.freeswitch.org/20853 >> >> Best regards, >> ?rn >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130501/76a45db4/attachment.html From mike at jerris.com Wed May 1 16:30:59 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 May 2013 08:30:59 -0400 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: References: Message-ID: Why don't we have a like button on this list? On Apr 30, 2013, at 10:28 PM, Cal Leeming [Simplicity Media Ltd] wrote: > I got 99 patents but SIP ain't one. > > Cal > > On Tue, Apr 30, 2013 at 2:06 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello, > > Thought I'd share this with you all... > > So there I was, quietly enjoying myself on theregister, when I stumbled upon what can only be described, as a complete and utter fail. > > http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ > > Responses so far? > > <@SwK> lololol > * foxx2 facepalm > BT come up with a regular stream of patents with no basis in reality > they shouldn't be able to patent a standard protocol > > I'm fairly sure my immediate response would be to print out 50 copies of [1] and send it via recorded delivery. > > It would be interesting to hear from anyone who has already been contacted by BT about this, and how they plan to deal with it. > > I've had enough of the internet for one day ;/ > > Cal > > [1] http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfwZk1r8alg0o1_500.png > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/0f1643fb/attachment.html From andrew at cassidywebservices.co.uk Wed May 1 16:42:49 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 1 May 2013 13:42:49 +0100 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: References: Message-ID: I thought the convention was to reply '+1' with no other text? :P On 1 May 2013 13:30, Michael Jerris wrote: > Why don't we have a like button on this list? > > On Apr 30, 2013, at 10:28 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > > I got 99 patents but SIP ain't one. > > Cal > > On Tue, Apr 30, 2013 at 2:06 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello, >> >> Thought I'd share this with you all... >> >> So there I was, quietly enjoying myself on theregister, when I stumbled >> upon what can only be described, as a complete and utter fail. >> >> >> http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ >> >> Responses so far? >> >> <@SwK> lololol >> * foxx2 facepalm >> BT come up with a regular stream of patents with no basis in >> reality >> they shouldn't be able to patent a standard protocol >> >> I'm fairly sure my immediate response would be to print out 50 copies of >> [1] and send it via recorded delivery. >> >> It would be interesting to hear from anyone who has already been >> contacted by BT about this, and how they plan to deal with it. >> >> I've had enough of the internet for one day ;/ >> >> Cal >> >> [1] >> http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfwZk1r8alg0o1_500.png >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *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/20130501/b9bb58f4/attachment-0001.html From intralanman at freeswitch.org Wed May 1 17:02:34 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Wed, 01 May 2013 09:02:34 -0400 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: References: Message-ID: <5181126A.3060204@freeswitch.org> +1 On 13-05-01 08:42 AM, Andrew Cassidy wrote: > I thought the convention was to reply '+1' with no other text? :P > > > On 1 May 2013 13:30, Michael Jerris > wrote: > > Why don't we have a like button on this list? > > On Apr 30, 2013, at 10:28 PM, Cal Leeming [Simplicity Media Ltd] > > wrote: > >> I got 99 patents but SIP ain't one. >> >> Cal >> >> On Tue, Apr 30, 2013 at 2:06 AM, Cal Leeming [Simplicity Media >> Ltd] > > wrote: >> >> Hello, >> >> Thought I'd share this with you all... >> >> So there I was, quietly enjoying myself on theregister, when >> I stumbled upon what can only be described, as a complete and >> utter fail. >> >> http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ >> >> Responses so far? >> >> <@SwK> lololol >> * foxx2 facepalm >> BT come up with a regular stream of patents with no >> basis in reality >> they shouldn't be able to patent a standard protocol >> >> I'm fairly sure my immediate response would be to print out >> 50 copies of [1] and send it via recorded delivery. >> >> It would be interesting to hear from anyone who has already >> been contacted by BT about this, and how they plan to deal >> with it. >> >> I've had enough of the internet for one day ;/ >> >> Cal >> >> [1] >> http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfwZk1r8alg0o1_500.png >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > *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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130501/08022185/attachment.html From abaci64 at gmail.com Wed May 1 18:08:35 2013 From: abaci64 at gmail.com (Abaci) Date: Wed, 01 May 2013 10:08:35 -0400 Subject: [Freeswitch-users] User settings in directory for unauthenticated users In-Reply-To: References: Message-ID: <518121E3.3000205@gmail.com> I have a request on Jira for an option to enable this http://jira.freeswitch.org/browse/FS-5009 (see the last few comments) On 5/1/2013 5:26 AM, Michel Brabants wrote: > > It works for me if I remember correctly. You should have the > cidr-attribute set on the user and the ip is then used to match it > with the user. > > Michel > > Op 1-mei-2013 04:43 schreef "Siri MM" > het volgende: > > Hi, > I have a system where I use Open ACL, and allow users to register > without any passwords required (one reason being all are trusted > devices withing the closed network) > However, I would like to restrict outgoing calls, and was hoping > to use the 'toll_allow' variable, and have set the same for the > user in the directory. However, in the dialplan, I see that > toll_allow is not set for the extension - when I experimented > by removing the open acl and forcing authentication, toll_allow is > set. > My question is, if this is expected behavior, is there any other > way I could pass on user specific settings to the dialplan in an > OpenACL environment? > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130501/fb3310b6/attachment.html From dwilkie at gmail.com Wed May 1 18:43:10 2013 From: dwilkie at gmail.com (David Wilkie) Date: Wed, 1 May 2013 21:43:10 +0700 Subject: [Freeswitch-users] Early Media with park () Message-ID: I'm using Adhearsion to bridge calls together which results in running the following command in FreeSwitch: bgapi originate {return_ring_ready=true,origination_uuid=c7f279eb-eec8-458a-b6e4-89ee1018567f,origination_caller_id_number='+14152345678,originate_timeout=30}sofia/gateway/didlogic/85512345678 &park() Leg B's (in this case 85512345678) phone rings but I cant hear any ringback signal. I tried this out in another FreeSwitch dialplan using bridge e.g. And I can hear the ringback tone, so I'm assuming it has something to to with the park(). Does anyone know how to get the ringback signal to work in the above case like Adhearsion generates? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/cae2d281/attachment-0001.html From msc at freeswitch.org Wed May 1 19:00:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 May 2013 08:00:54 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Message-ID: We look forward to our call today: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_01 We have Omar from OrecX who will be discussing call recording systems. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/79fa776e/attachment.html From chris at ghosttelecom.com Wed May 1 19:27:09 2013 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 1 May 2013 15:27:09 +0000 Subject: [Freeswitch-users] Retransmission resolves new ip address Message-ID: <4A3FDB652E75BE4AAFD7CB1FB45FF55B0FA7263C@DB3PRD0710MB392.eurprd07.prod.outlook.com> Hi, Have a scenario whereby FS sends an invite to another downstream sip server which is late replying and fs then sends a re-transmission, however fs then subsequently gets a reply. The problem I have is that the downstream server is on a dns-srv roundrobin between 2 servers and the re-transmission is resolving a different server. This seems to be causing the replies that have subsequently come in from the original server to just bounce until they timeout. The re-transmission to the other server seems to carry on fine but sometimes generates a 482 merged call error. The main effect is just messy call stats as I get strange 483 errors generated by fs for some reason internally for the first call and also a normal error or cdr for the re-transmitted part of the call. If I set the downstream server to a fixed ip everything works fine. Should the retransmission do a new dns lookup? If it gets a new ip address should fs handle the dialog on the old ip address? I may be just not setting something correctly but cannot find any references to this problem in any blogs, any ideas. Many thanks Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/2e55a2f6/attachment.html From itsusama at gmail.com Wed May 1 19:31:43 2013 From: itsusama at gmail.com (Usama Zaidi) Date: Wed, 1 May 2013 20:31:43 +0500 Subject: [Freeswitch-users] BT claims licensing on SIP Message-ID: <008101ce4680$ff1a3970$fd4eac50$@gmail.com> +1 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of freeswitch-users-request at lists.freeswitch.org Sent: Wednesday, May 01, 2013 7:44 PM To: freeswitch-users at lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 83, Issue 4 Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." From grcamauer at gmail.com Wed May 1 19:37:28 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 1 May 2013 12:37:28 -0300 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: <008101ce4680$ff1a3970$fd4eac50$@gmail.com> References: <008101ce4680$ff1a3970$fd4eac50$@gmail.com> Message-ID: <6395963092579871130@unknownmsgid> The news is on Slashdot now. Sent from my iPhone On 01/05/2013, at 12:34, Usama Zaidi wrote: > +1 > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > freeswitch-users-request at lists.freeswitch.org > Sent: Wednesday, May 01, 2013 7:44 PM > To: freeswitch-users at lists.freeswitch.org > Subject: FreeSWITCH-users Digest, Vol 83, Issue 4 > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific than > "Re: Contents of FreeSWITCH-users digest..." > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andrew at cassidywebservices.co.uk Wed May 1 19:52:42 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 1 May 2013 16:52:42 +0100 Subject: [Freeswitch-users] BT claims licensing on SIP In-Reply-To: <6395963092579871130@unknownmsgid> References: <008101ce4680$ff1a3970$fd4eac50$@gmail.com> <6395963092579871130@unknownmsgid> Message-ID: +1 On 1 May 2013 16:37, Guillermo Ruiz Camauer wrote: > The news is on Slashdot now. > > Sent from my iPhone > > On 01/05/2013, at 12:34, Usama Zaidi wrote: > > > +1 > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > freeswitch-users-request at lists.freeswitch.org > > Sent: Wednesday, May 01, 2013 7:44 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: FreeSWITCH-users Digest, Vol 83, Issue 4 > > > > Send FreeSWITCH-users mailing list submissions to > > freeswitch-users at lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request at lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner at lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific than > > "Re: Contents of FreeSWITCH-users digest..." > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- *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/20130501/553d237b/attachment-0001.html From marketing at cluecon.com Thu May 2 00:44:43 2013 From: marketing at cluecon.com (Michael Collins) Date: Wed, 1 May 2013 13:44:43 -0700 Subject: [Freeswitch-users] Erlang training at ClueCon 2013 - would you be interested? Message-ID: Hello all! As you know, ClueCon 2013 is coming up fast. Lots of things are happening behind the scenes, including preparations for possible pre- and post-cluecon training sessions. One topic of consideration is Erlang. We want to gauge the interest level in having a training session on Erlang. If you are definitely interested in attending an Erlang training session at ClueCon 2013 please let us know. Reply off-list to marketing at cluecon.com and CC james at 2600hz.org. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/cbca55f2/attachment.html From jaybinks at gmail.com Thu May 2 02:35:33 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 2 May 2013 08:35:33 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: Message-ID: So what would one put in sip_execute_on_image to make FS knock back a T38 Re-invite ? :) Respond with an SDP, with no T38 I guess.. On 1 May 2013 01:28, Anthony Minessale wrote: > The variable is sip_execute_on_image > > > > On Tue, Apr 30, 2013 at 5:35 AM, ?rn Arnarson wrote: > >> Hello, >> >> I'm trying to get T38 gatewaying to work. I have a call coming in with >> FAX audio, and an endpoint that needs T38. The endpoint will send a T38 >> re-invite, to which FS responds 100 trying, and after 40 seconds the >> endpoint will give up (citing stale 100 trying). >> >> I set these variables before briding: >> >> >> >> And then I bridge like this (dialstring is sofia/profile/number at ip:port): >> >> >> I've tried all variants I can think of, and always have the same result, >> except for when fax_enable_t38 = false, in which case FS rejects the >> re-invite. >> >> I found some old discussions on the mailing list alluding to the same >> problem, but not much in the way of solutions. >> >> Can anybody give me a pointer as to what I'm doing wrong? I was running a >> git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the same >> results. >> >> Here's what happens in FS. At the very end you can see the Re-INVITE, and >> then nothing happens post that point. >> http://pastebin.freeswitch.org/20853 >> >> Best regards, >> ?rn >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/8ab41104/attachment.html From anthony.minessale at gmail.com Thu May 2 02:54:40 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 May 2013 17:54:40 -0500 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: Message-ID: its to make the call xfer to rxfax or to t38_gateway app when it sees t38 re-invite. Its probably on the wiki. On Wed, May 1, 2013 at 5:35 PM, jay binks wrote: > So what would one put in sip_execute_on_image > to make FS knock back a T38 Re-invite ? :) > > Respond with an SDP, with no T38 I guess.. > > > On 1 May 2013 01:28, Anthony Minessale wrote: > >> The variable is sip_execute_on_image >> >> >> >> On Tue, Apr 30, 2013 at 5:35 AM, ?rn Arnarson wrote: >> >>> Hello, >>> >>> I'm trying to get T38 gatewaying to work. I have a call coming in with >>> FAX audio, and an endpoint that needs T38. The endpoint will send a T38 >>> re-invite, to which FS responds 100 trying, and after 40 seconds the >>> endpoint will give up (citing stale 100 trying). >>> >>> I set these variables before briding: >>> >>> >>> >>> And then I bridge like this (dialstring is sofia/profile/number at ip >>> :port): >>> >>> >>> I've tried all variants I can think of, and always have the same result, >>> except for when fax_enable_t38 = false, in which case FS rejects the >>> re-invite. >>> >>> I found some old discussions on the mailing list alluding to the same >>> problem, but not much in the way of solutions. >>> >>> Can anybody give me a pointer as to what I'm doing wrong? I was running >>> a git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the >>> same results. >>> >>> Here's what happens in FS. At the very end you can see the Re-INVITE, >>> and then nothing happens post that point. >>> http://pastebin.freeswitch.org/20853 >>> >>> Best regards, >>> ?rn >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/d5f474bc/attachment-0001.html From bedgar at vseinc.com Thu May 2 03:39:08 2013 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Wed, 1 May 2013 19:39:08 -0400 Subject: [Freeswitch-users] freetdm RBS configuration for multi-port cards In-Reply-To: <333789DE5C38474EB3A478A538F4EBAB35BCB5D34A@prod-exch01.corp.vseinc.com> References: <333789DE5C38474EB3A478A538F4EBAB35BCB5D32A@prod-exch01.corp.vseinc.com> <333789DE5C38474EB3A478A538F4EBAB35BCB5D348@prod-exch01.corp.vseinc.com> <333789DE5C38474EB3A478A538F4EBAB35BCB5D34A@prod-exch01.corp.vseinc.com> Message-ID: <31A79B0B4414EB4B83C7EAF307ED57DA0351834C@prod-exch01.corp.vseinc.com> Moises, I recreated the core dump and opened the Jira as you requested. Thank you, Brian From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Edgar Sent: Saturday, April 20, 2013 9:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] freetdm RBS configuration for multi-port cards Moises, Thank you. I will reproduce the core dump and will post and assign as requested. Regards, Brian C. Edgar, Jr. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva Sent: Saturday, April 20, 2013 8:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freetdm RBS configuration for multi-port cards On Sat, Apr 20, 2013 at 5:30 PM, > wrote: Moises, Thanks for the reply. I did do that and FS core dumped when I cycled loopback in my mux on several T1?s so I thought that I had some configuration errors. I am using 2x4 port Sangoma PCI-Express cards. We may have to investigate mod_freetdm from the recent development effort we invested in. We are using: FreeSWITCH Version 1.2.8+git~20130410T033224Z~8dd4c8f9f1 (git 8dd4c8f 2013-04-10 03:32:24Z) A core dump is never acceptable regardless of whether there is any configuration error. If you can reproduce the core dump follow instructions here to report the bug: http://wiki.freeswitch.org/wiki/Reporting_Bugs Assign the bug to me. Moises Silva Manager, Software Engineering msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/29a18b2a/attachment.html From jaybinks at gmail.com Thu May 2 04:07:56 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 2 May 2013 10:07:56 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: Message-ID: yup I saw that.... and I thought I already knew the answer to that question. but im still yet to find a way to specifically knock back a T38 re-invite :( ( FS does this 100 trying... but nothing further .. ) On 2 May 2013 08:54, Anthony Minessale wrote: > its to make the call xfer to rxfax or to t38_gateway app when it sees t38 > re-invite. > Its probably on the wiki. > > > > On Wed, May 1, 2013 at 5:35 PM, jay binks wrote: > >> So what would one put in sip_execute_on_image >> to make FS knock back a T38 Re-invite ? :) >> >> Respond with an SDP, with no T38 I guess.. >> >> >> On 1 May 2013 01:28, Anthony Minessale wrote: >> >>> The variable is sip_execute_on_image >>> >>> >>> >>> On Tue, Apr 30, 2013 at 5:35 AM, ?rn Arnarson wrote: >>> >>>> Hello, >>>> >>>> I'm trying to get T38 gatewaying to work. I have a call coming in with >>>> FAX audio, and an endpoint that needs T38. The endpoint will send a T38 >>>> re-invite, to which FS responds 100 trying, and after 40 seconds the >>>> endpoint will give up (citing stale 100 trying). >>>> >>>> I set these variables before briding: >>>> >>>> >>>> >>>> And then I bridge like this (dialstring is sofia/profile/number at ip >>>> :port): >>>> >>>> >>>> I've tried all variants I can think of, and always have the same >>>> result, except for when fax_enable_t38 = false, in which case FS rejects >>>> the re-invite. >>>> >>>> I found some old discussions on the mailing list alluding to the same >>>> problem, but not much in the way of solutions. >>>> >>>> Can anybody give me a pointer as to what I'm doing wrong? I was running >>>> a git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the >>>> same results. >>>> >>>> Here's what happens in FS. At the very end you can see the Re-INVITE, >>>> and then nothing happens post that point. >>>> http://pastebin.freeswitch.org/20853 >>>> >>>> Best regards, >>>> ?rn >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/f4759850/attachment-0001.html From brian at freeswitch.org Thu May 2 06:23:44 2013 From: brian at freeswitch.org (Brian West) Date: Wed, 1 May 2013 21:23:44 -0500 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: Message-ID: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> what is the entire call flow? And what rev are you using... fax_enable_t38_request=true will make us send a re-invite. On May 1, 2013, at 7:07 PM, jay binks wrote: > yup I saw that.... and I thought I already knew the answer to that question. > but im still yet to find a way to specifically knock back a T38 re-invite :( > > ( FS does this 100 trying... but nothing further .. ) > > From jaybinks at gmail.com Thu May 2 07:00:36 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 2 May 2013 13:00:36 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: I have tested this with current versions... ok so unlike many others... I find G711 gives better faxing than T.38 ( for many reasons... but mainly because all my network is fibre and I control a lot of it myself ). I have one carrier that that insists on doing fax tone detection in their SBC. the call comes in G711... they detect fax tones then re-invite T.38. I want to knock the T.38 back, but cant.. if I disable T.38 we get the 100 Trying... then nothing further. only way I can inter-op is to use t38. On 2 May 2013 12:23, Brian West wrote: > what is the entire call flow? And what rev are you using... > fax_enable_t38_request=true will make us send a re-invite. > > On May 1, 2013, at 7:07 PM, jay binks wrote: > > > yup I saw that.... and I thought I already knew the answer to that > question. > > but im still yet to find a way to specifically knock back a T38 > re-invite :( > > > > ( FS does this 100 trying... but nothing further .. ) > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/1078dbc2/attachment.html From anthony.minessale at gmail.com Thu May 2 08:17:54 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 1 May 2013 23:17:54 -0500 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: Clarify what knock back means. REFUSE it? or SEND IT? On Wed, May 1, 2013 at 10:00 PM, jay binks wrote: > I have tested this with current versions... > > ok so unlike many others... I find G711 gives better faxing than T.38 > ( for many reasons... but mainly because all my network is fibre and I > control a lot of it myself ). > > I have one carrier that that insists on doing fax tone detection in their > SBC. > the call comes in G711... they detect fax tones then re-invite T.38. > > I want to knock the T.38 back, but cant.. > if I disable T.38 we get the 100 Trying... then nothing further. > only way I can inter-op is to use t38. > > > > > On 2 May 2013 12:23, Brian West wrote: > >> what is the entire call flow? And what rev are you using... >> fax_enable_t38_request=true will make us send a re-invite. >> >> On May 1, 2013, at 7:07 PM, jay binks wrote: >> >> > yup I saw that.... and I thought I already knew the answer to that >> question. >> > but im still yet to find a way to specifically knock back a T38 >> re-invite :( >> > >> > ( FS does this 100 trying... but nothing further .. ) >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130501/0eabcd51/attachment.html From jaybinks at gmail.com Thu May 2 10:16:59 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 2 May 2013 16:16:59 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: refuse it.. if I get a T.38 invite... I want to go "nah man.. G711 only" On 2 May 2013 14:17, Anthony Minessale wrote: > Clarify what knock back means. REFUSE it? or SEND IT? > > > On Wed, May 1, 2013 at 10:00 PM, jay binks wrote: > >> I have tested this with current versions... >> >> ok so unlike many others... I find G711 gives better faxing than T.38 >> ( for many reasons... but mainly because all my network is fibre and I >> control a lot of it myself ). >> >> I have one carrier that that insists on doing fax tone detection in their >> SBC. >> the call comes in G711... they detect fax tones then re-invite T.38. >> >> I want to knock the T.38 back, but cant.. >> if I disable T.38 we get the 100 Trying... then nothing further. >> only way I can inter-op is to use t38. >> >> >> >> >> On 2 May 2013 12:23, Brian West wrote: >> >>> what is the entire call flow? And what rev are you using... >>> fax_enable_t38_request=true will make us send a re-invite. >>> >>> On May 1, 2013, at 7:07 PM, jay binks wrote: >>> >>> > yup I saw that.... and I thought I already knew the answer to that >>> question. >>> > but im still yet to find a way to specifically knock back a T38 >>> re-invite :( >>> > >>> > ( FS does this 100 trying... but nothing further .. ) >>> > >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/f886c872/attachment-0001.html From navnath.sonavne at yahoo.com Thu May 2 14:32:16 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Thu, 2 May 2013 18:32:16 +0800 (SGT) Subject: [Freeswitch-users] Make call from Ekiga phone via freeswitch Message-ID: <1367490736.13066.YahooMailNeo@web192205.mail.sg3.yahoo.com> Hi All, I am able to establish audio call from X-Lite(sip protocol) soft phone? to Ekiga(h323 protocol)?softphone?via freeswitch. I can hear voice from both sides. Now this call making is one way i.e. i can make call only? from X-Lite?softphone?because it is registered on freeswitch. Now I want to make call from Ekiga?softphone?to X-Lite?softphone via freeswitch. My first query is How to register Ekiga phone on freeswitch? Once phone is registered on freeswitch How to make call to X-Lite phone? -Navnath. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/0cd3b06d/attachment.html From bdfoster at davri.com Thu May 2 14:59:34 2013 From: bdfoster at davri.com (Brian Foster) Date: Thu, 2 May 2013 06:59:34 -0400 Subject: [Freeswitch-users] Make call from Ekiga phone via freeswitch In-Reply-To: <1367490736.13066.YahooMailNeo@web192205.mail.sg3.yahoo.com> References: <1367490736.13066.YahooMailNeo@web192205.mail.sg3.yahoo.com> Message-ID: /usr/local/freeswitch/conf/sip_profiles/internal.xml: Set multiple-registrations to true Re start FS or reloadxml and internal profile. On May 2, 2013 6:38 AM, "Navnath Sonavne" wrote: > Hi All, > > I am able to establish audio call from X-Lite(sip protocol) soft phone > to Ekiga(h323 protocol) softphone via freeswitch. > I can hear voice from both sides. > Now this call making is one way i.e. i can make call only > from X-Lite softphone because it is registered on freeswitch. > Now I want to make call from Ekiga softphone to X-Lite softphone via > freeswitch. > My first query is How to register Ekiga phone on freeswitch? > Once phone is registered on freeswitch How to make call to X-Lite phone? > > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/5051fb4f/attachment.html From smfarrukh at live.com Thu May 2 15:04:32 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Thu, 2 May 2013 11:04:32 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130502/2dbc414b/attachment.html From bdfoster at davri.com Thu May 2 16:51:42 2013 From: bdfoster at davri.com (Brian Foster) Date: Thu, 2 May 2013 08:51:42 -0400 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: > Thanks everyone, and Steve timers might not be efficient solution but for > now it will be good for testing purpose, could you please tell me exactly > which file and parameter should I look for to change, is this configuration > located in sofia.conf.xml ? kindly reply! > and thanks to all again! > > Regards, > Muhammad Farrukh > > ------------------------------ > From: steveayre at gmail.com > Date: Tue, 30 Apr 2013 17:20:36 +0100 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > SIP over UDP has no way of knowing the other side has lost connectivity > (eg wifi down). Sip droid will know the local network is down and be able > to hang up the call because of that. FreeSWITCH has no such knowledge, so > won't know the call is hung up and will continue sending RTP. > > 3 possibilities are: > 1) Use session timers > 2) There is a parameter to hang up a call is no RTP has been received for > a certain period of time > 3) Use SIP/TCP, keepalives might help (but the default is measured in > hours) > > Note that all of these are timeouts - you simply won't be able to know the > call has ended immediately. > > Steve > > > > On 30 Apr 2013, at 08:33, Farrukh Ali wrote: > > Hi, > > I need some help, there is an issue in Freeswitch with SIP clients, when a > call is not hung up properly Freeswitch does not close its RTP channels, > and it starts to get load on bandwidth after too many dead channels are > generated, Improper hang up means if during call sip client went out of the > range of WiFi, the client i.e SipDroid shows call hang up but freeswitch > continues to send voice packets. Kindly reply > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/9dc4e9c9/attachment-0001.html From navnath.sonavne at yahoo.com Thu May 2 16:58:59 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Thu, 2 May 2013 20:58:59 +0800 (SGT) Subject: [Freeswitch-users] make ekiga to flex call Message-ID: <1367499539.94981.YahooMailNeo@web192203.mail.sg3.yahoo.com> Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source? to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension.? After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes? in default.xml accordingly to forward calls to this ip. Here is My default.xml part : ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Ekiga phone can answer the call and there is audio?transmission?from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to?flex?client. How to call flex client from ekiga softphone? Anybody?please?me on this issue. -Navnath. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/8b30ca51/attachment.html From bdfoster at davri.com Thu May 2 17:18:03 2013 From: bdfoster at davri.com (Brian Foster) Date: Thu, 2 May 2013 09:18:03 -0400 Subject: [Freeswitch-users] make ekiga to flex call In-Reply-To: <1367499539.94981.YahooMailNeo@web192203.mail.sg3.yahoo.com> References: <1367499539.94981.YahooMailNeo@web192203.mail.sg3.yahoo.com> Message-ID: According to your dialplan you're forcing all calls that hit Local_Extension (any call with a destination number of 4 digits) to call your ekiga softphone. Your other problem lies in the fact that you should have different dialplans for different methods of calling unlike endpoints one for h232 one for flex. See below: -BDF On May 2, 2013 9:05 AM, "Navnath Sonavne" wrote: > > Hi All, > > I am using flex client(at 192.168.9.165) given in freeswitch source > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). > I have two users registered on freeswitch with extension as 1100 and 1101 in default context. > I loged in using one of extension in flex client,then I dail another extension. > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes > in default.xml accordingly to forward calls to this ip. > Here is My default.xml part : > > > > > > < > > > < > > > Ekiga phone can answer the call and there is audio transmission from both end successfully. > > But this is one way calling i.e. from flex to ekiga. > Now I want to call from Ekiga softphone to flex client. > How to call flex client from ekiga softphone? > > Anybody please me on this issue. > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/d0bb2ee7/attachment.html From mehroz.ashraf85 at gmail.com Thu May 2 17:49:48 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 2 May 2013 06:49:48 -0700 (PDT) Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> Message-ID: <1367502588238-7590307.post@n2.nabble.com> What if we want to even hungup the channels associated with those users? does freeswitch helps ? Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu May 2 18:34:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 2 May 2013 09:34:39 -0500 Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: <1367502588238-7590307.post@n2.nabble.com> References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> <1367502588238-7590307.post@n2.nabble.com> Message-ID: <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers Ken Sent from my iPad On May 2, 2013, at 8:49, mehroz wrote: > What if we want to even hungup the channels associated with those users? does > freeswitch helps ? > > Like, i have a scenario in which , an active call disrupts , once BOTH users > gets disconnected from network. > Here, both users gets unregistered after option messages but the call > sustains. Is there any thing i can handle this scenario? > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.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://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 From mehroz.ashraf85 at gmail.com Thu May 2 19:13:29 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 2 May 2013 08:13:29 -0700 (PDT) Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> <1367502588238-7590307.post@n2.nabble.com> <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> Message-ID: <1367507609182-7590309.post@n2.nabble.com> i cannot use session-timers, disabled due to an issue with video calling.... and rtp timers just does not seems to be working! i have But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call ..... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu May 2 19:28:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 May 2013 08:28:29 -0700 Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: <1367507609182-7590309.post@n2.nabble.com> References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> <1367502588238-7590307.post@n2.nabble.com> <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> <1367507609182-7590309.post@n2.nabble.com> Message-ID: On Thu, May 2, 2013 at 8:13 AM, mehroz wrote: > i cannot use session-timers, disabled due to an issue with video > calling.... > > and rtp timers just does not seems to be working! > Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected? > i have > > But once network disconnects from both clients (works good, with a single > client network outage), FS does not hangs up the call ..... > Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/d8ecd190/attachment-0001.html From anthony.minessale at gmail.com Thu May 2 19:54:43 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 May 2013 10:54:43 -0500 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: I'm on my cell so I can't look but its like refuse_t38=true or something similar on the leg facing the t38 On May 2, 2013 1:21 AM, "jay binks" wrote: > refuse it.. > > if I get a T.38 invite... I want to go "nah man.. G711 only" > > > > On 2 May 2013 14:17, Anthony Minessale wrote: > >> Clarify what knock back means. REFUSE it? or SEND IT? >> >> >> On Wed, May 1, 2013 at 10:00 PM, jay binks wrote: >> >>> I have tested this with current versions... >>> >>> ok so unlike many others... I find G711 gives better faxing than T.38 >>> ( for many reasons... but mainly because all my network is fibre and I >>> control a lot of it myself ). >>> >>> I have one carrier that that insists on doing fax tone detection in >>> their SBC. >>> the call comes in G711... they detect fax tones then re-invite T.38. >>> >>> I want to knock the T.38 back, but cant.. >>> if I disable T.38 we get the 100 Trying... then nothing further. >>> only way I can inter-op is to use t38. >>> >>> >>> >>> >>> On 2 May 2013 12:23, Brian West wrote: >>> >>>> what is the entire call flow? And what rev are you using... >>>> fax_enable_t38_request=true will make us send a re-invite. >>>> >>>> On May 1, 2013, at 7:07 PM, jay binks wrote: >>>> >>>> > yup I saw that.... and I thought I already knew the answer to that >>>> question. >>>> > but im still yet to find a way to specifically knock back a T38 >>>> re-invite :( >>>> > >>>> > ( FS does this 100 trying... but nothing further .. ) >>>> > >>>> > >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/0d871334/attachment.html From jmesquita at freeswitch.org Thu May 2 22:48:37 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 2 May 2013 15:48:37 -0300 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { I hope this helps... Jo?o Mesquita FreeSWITCH? Solutions On Thu, May 2, 2013 at 12:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I'm on my cell so I can't look but its like refuse_t38=true or something > similar on the leg facing the t38 > On May 2, 2013 1:21 AM, "jay binks" wrote: > >> refuse it.. >> >> if I get a T.38 invite... I want to go "nah man.. G711 only" >> >> >> >> On 2 May 2013 14:17, Anthony Minessale wrote: >> >>> Clarify what knock back means. REFUSE it? or SEND IT? >>> >>> >>> On Wed, May 1, 2013 at 10:00 PM, jay binks wrote: >>> >>>> I have tested this with current versions... >>>> >>>> ok so unlike many others... I find G711 gives better faxing than T.38 >>>> ( for many reasons... but mainly because all my network is fibre and I >>>> control a lot of it myself ). >>>> >>>> I have one carrier that that insists on doing fax tone detection in >>>> their SBC. >>>> the call comes in G711... they detect fax tones then re-invite T.38. >>>> >>>> I want to knock the T.38 back, but cant.. >>>> if I disable T.38 we get the 100 Trying... then nothing further. >>>> only way I can inter-op is to use t38. >>>> >>>> >>>> >>>> >>>> On 2 May 2013 12:23, Brian West wrote: >>>> >>>>> what is the entire call flow? And what rev are you using... >>>>> fax_enable_t38_request=true will make us send a re-invite. >>>>> >>>>> On May 1, 2013, at 7:07 PM, jay binks wrote: >>>>> >>>>> > yup I saw that.... and I thought I already knew the answer to that >>>>> question. >>>>> > but im still yet to find a way to specifically knock back a T38 >>>>> re-invite :( >>>>> > >>>>> > ( FS does this 100 trying... but nothing further .. ) >>>>> > >>>>> > >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Sincerely >>>> >>>> Jay >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/5e032de0/attachment-0001.html From tnsampaio at bsd.com.br Thu May 2 23:43:00 2013 From: tnsampaio at bsd.com.br (Tiago Sampaio) Date: Thu, 02 May 2013 16:43:00 -0300 Subject: [Freeswitch-users] Blind transfer with call waiting Message-ID: <5182C1C4.40804@bsd.com.br> Is there something like blind transfer with call waiting when the target is busy? I have a situation here, EX: When Jhon (user 1001) answer a call and and customer ask to talk to Robert (user 1002), Jhon do a blind tansfer an hangup. Currently im wasting that call when Robert is busy. I need an way to park the call and when Robert become available ring his extension (and maybe put an timeout and call back to Jhon). Im currently writing an script with perl to do this, something like that (i do this from memory): use strict; use POSIX qw(strftime); my ($target_ext,$source_ext,$tryes); my $max_time = 30; our $session; $target_ext = $session->getVariable('target_ext'); $source_ext = $session->getVariable('source_ext'); while($tryes <= $max_time){ my $uuid = $session->execute("db","select/in_a_call/$target_ext"); if(!$uuid) $session->execute("trasfer","$target_ext XML transfer_context"); sleep(1); $tryes++; } $session->execute("transfer","$source_ext XML transfer_context"); Before call an extension i do db insert/in_a_call//${uuid} and after db delete/in_a_call/ But in thinking will be too hard to maintain this code and must be an easier way to do this... Any help? From ibk at labhijau.net Thu May 2 23:38:08 2013 From: ibk at labhijau.net (Iwan Budi Kusnanto) Date: Fri, 3 May 2013 02:38:08 +0700 Subject: [Freeswitch-users] Copy rtmp extra header to sofia Message-ID: Hi, I want to copy RTMP extra header to Sofia/SIP. I can do it in dialplan like this This way, i need to specify each header name in dialplan. Can we make dialplan that support arbitrary header name? I can do it in Lua with this method: - execute uuid_dump - search line that started by variable_rtmp_u__X-Flash - extract header_name and header_value - session:execute("export", "nolocal:sip_h_" .. header_name .. "=" .. header_value) Any better way to do this? Thanks -- Iwan Budi Kusnanto From anthony.minessale at gmail.com Fri May 3 00:12:07 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 2 May 2013 15:12:07 -0500 Subject: [Freeswitch-users] Blind transfer with call waiting In-Reply-To: <5182C1C4.40804@bsd.com.br> References: <5182C1C4.40804@bsd.com.br> Message-ID: you could try http://wiki.freeswitch.org/wiki/Variable_campon On Thu, May 2, 2013 at 2:43 PM, Tiago Sampaio wrote: > Is there something like blind transfer with call waiting when the target > is busy? > > I have a situation here, EX: When Jhon (user 1001) answer a call and and > customer ask to talk to Robert (user 1002), > Jhon do a blind tansfer an hangup. Currently im wasting that call when > Robert is busy. I need an way to park the > call and when Robert become available ring his extension (and maybe put > an timeout and call back to Jhon). > > Im currently writing an script with perl to do this, something like that > (i do this from memory): > > use strict; > use POSIX qw(strftime); > my ($target_ext,$source_ext,$tryes); > my $max_time = 30; > > our $session; > > $target_ext = $session->getVariable('target_ext'); > $source_ext = $session->getVariable('source_ext'); > > while($tryes <= $max_time){ > my $uuid = $session->execute("db","select/in_a_call/$target_ext"); > > if(!$uuid) > $session->execute("trasfer","$target_ext XML transfer_context"); > > sleep(1); > $tryes++; > } > > $session->execute("transfer","$source_ext XML transfer_context"); > > > Before call an extension i do db insert/in_a_call//${uuid} > and after db delete/in_a_call/ > > But in thinking will be too hard to maintain this code and must be an > easier way to do this... > > Any help? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/9c212581/attachment.html From msc at freeswitch.org Fri May 3 00:22:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 May 2013 13:22:43 -0700 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: > if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { > Thanks for doing that. It appears that our T38 content on the wiki is woefully thin: http://wiki.freeswitch.org/wiki/T.38 If anyone out there has T.38 experience/knowledge/battle scars and they are willing to share that would be most appreciated. If you have specific configurations that you can post we could start with that. Also, if anyone has logs and/or SIP captures of T.38 exchanges and would be willing to post those please do so, or contact me off list and we'll talk about getting this item some more documentation. -MC > > I hope this helps... > > Jo?o Mesquita > FreeSWITCH? Solutions > > -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/1178587d/attachment.html From jmesquita at freeswitch.org Fri May 3 00:37:19 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 2 May 2013 17:37:19 -0300 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: Configuring FreeSWITCH to support T.38 is really easy and I can help with that. On the other hand, debugging T.38 communications is pretty hard since I have no idea how T.38 works, lol. Jo?o Mesquita FreeSWITCH? Solutions On Thu, May 2, 2013 at 5:22 PM, Michael Collins wrote: > > > > On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: > >> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >> > > Thanks for doing that. It appears that our T38 content on the wiki is > woefully thin: > http://wiki.freeswitch.org/wiki/T.38 > > If anyone out there has T.38 experience/knowledge/battle scars and they > are willing to share that would be most appreciated. If you have specific > configurations that you can post we could start with that. Also, if anyone > has logs and/or SIP captures of T.38 exchanges and would be willing to post > those please do so, or contact me off list and we'll talk about getting > this item some more documentation. > -MC > > >> >> I hope this helps... >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> -MC > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/bdb1fe9a/attachment-0001.html From msc at freeswitch.org Fri May 3 00:42:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 May 2013 13:42:40 -0700 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: On Thu, May 2, 2013 at 1:37 PM, Jo?o Mesquita wrote: > Configuring FreeSWITCH to support T.38 is really easy and I can help with > that. On the other hand, debugging T.38 communications is pretty hard since > I have no idea how T.38 works, lol. > Yep! that's the issue in a nutshell. -MC > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Thu, May 2, 2013 at 5:22 PM, Michael Collins wrote: > >> >> >> >> On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: >> >>> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >>> >> >> Thanks for doing that. It appears that our T38 content on the wiki is >> woefully thin: >> http://wiki.freeswitch.org/wiki/T.38 >> >> If anyone out there has T.38 experience/knowledge/battle scars and they >> are willing to share that would be most appreciated. If you have specific >> configurations that you can post we could start with that. Also, if anyone >> has logs and/or SIP captures of T.38 exchanges and would be willing to post >> those please do so, or contact me off list and we'll talk about getting >> this item some more documentation. >> -MC >> >> >>> >>> I hope this helps... >>> >>> Jo?o Mesquita >>> FreeSWITCH? Solutions >>> >>> -MC >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/b4914fea/attachment.html From adahary at gmail.com Fri May 3 01:13:55 2013 From: adahary at gmail.com (adahary) Date: Thu, 2 May 2013 14:13:55 -0700 (PDT) Subject: [Freeswitch-users] Client TLS certificate setup Message-ID: <1367529235228-7590319.post@n2.nabble.com> I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well.I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this:tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sipstport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATINGtport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sipsI will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients.Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-tp7590319.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130502/28b013d5/attachment.html From jaybinks at gmail.com Fri May 3 05:43:16 2013 From: jaybinks at gmail.com (jay binks) Date: Fri, 3 May 2013 11:43:16 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: im 90% sure I tried that... I trolled the source looking for all the hidden gems .. ill have to lab it up again... but I dont believe refuse_t38 responded to the T.38 invite at all. On 3 May 2013 06:22, Michael Collins wrote: > > > > On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: > >> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >> > > Thanks for doing that. It appears that our T38 content on the wiki is > woefully thin: > http://wiki.freeswitch.org/wiki/T.38 > > If anyone out there has T.38 experience/knowledge/battle scars and they > are willing to share that would be most appreciated. If you have specific > configurations that you can post we could start with that. Also, if anyone > has logs and/or SIP captures of T.38 exchanges and would be willing to post > those please do so, or contact me off list and we'll talk about getting > this item some more documentation. > -MC > > >> >> I hope this helps... >> >> Jo?o Mesquita >> FreeSWITCH? Solutions >> >> -MC > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/6b24a603/attachment.html From jmesquita at freeswitch.org Fri May 3 06:14:03 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 2 May 2013 23:14:03 -0300 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: If it didn't we would be happy to respond to the Jira opened! :) Sent from my iPhone On May 2, 2013, at 10:43 PM, jay binks wrote: > im 90% sure I tried that... > I trolled the source looking for all the hidden gems .. > > ill have to lab it up again... but I dont believe refuse_t38 responded to the T.38 invite at all. > > > > > > > On 3 May 2013 06:22, Michael Collins wrote: >> >> >> >> On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: >>> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >> >> Thanks for doing that. It appears that our T38 content on the wiki is woefully thin: >> http://wiki.freeswitch.org/wiki/T.38 >> >> If anyone out there has T.38 experience/knowledge/battle scars and they are willing to share that would be most appreciated. If you have specific configurations that you can post we could start with that. Also, if anyone has logs and/or SIP captures of T.38 exchanges and would be willing to post those please do so, or contact me off list and we'll talk about getting this item some more documentation. >> -MC >> >>> >>> I hope this helps... >>> >>> Jo?o Mesquita >>> FreeSWITCH? Solutions >> -MC >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > -- > Sincerely > > Jay > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/e712000f/attachment-0001.html From spencer at 5ninesolutions.com Fri May 3 06:24:51 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 2 May 2013 21:24:51 -0500 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: <616A275F-8209-44AE-880D-50259C24AAF1@5ninesolutions.com> I had this exact problem after an upgrade. The problem that the t38-passthru Sofia profile param is ignored on latest. Set t38_passthru as a global variable and give that a shot. I'm working on collecting logs for a jira now. Spencer Sent from my iPad On May 2, 2013, at 9:14 PM, Jo?o Mesquita wrote: > If it didn't we would be happy to respond to the Jira opened! :) > > Sent from my iPhone > > On May 2, 2013, at 10:43 PM, jay binks wrote: > >> im 90% sure I tried that... >> I trolled the source looking for all the hidden gems .. >> >> ill have to lab it up again... but I dont believe refuse_t38 responded to the T.38 invite at all. >> >> >> >> >> >> >> On 3 May 2013 06:22, Michael Collins wrote: >>> >>> >>> >>> On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: >>>> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >>> >>> Thanks for doing that. It appears that our T38 content on the wiki is woefully thin: >>> http://wiki.freeswitch.org/wiki/T.38 >>> >>> If anyone out there has T.38 experience/knowledge/battle scars and they are willing to share that would be most appreciated. If you have specific configurations that you can post we could start with that. Also, if anyone has logs and/or SIP captures of T.38 exchanges and would be willing to post those please do so, or contact me off list and we'll talk about getting this item some more documentation. >>> -MC >>> >>>> >>>> I hope this helps... >>>> >>>> Jo?o Mesquita >>>> FreeSWITCH? Solutions >>> -MC >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >> >> -- >> Sincerely >> >> Jay >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130502/39e82afb/attachment.html From jaybinks at gmail.com Fri May 3 07:06:41 2013 From: jaybinks at gmail.com (jay binks) Date: Fri, 3 May 2013 13:06:41 +1000 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: yea yea .. I know I already have enought jira's opened. I cant keep up with providing the feedback to bugs I submit :P not enough hours in the day. On 3 May 2013 12:14, Jo?o Mesquita wrote: > If it didn't we would be happy to respond to the Jira opened! :) > > Sent from my iPhone > > On May 2, 2013, at 10:43 PM, jay binks wrote: > > im 90% sure I tried that... > I trolled the source looking for all the hidden gems .. > > ill have to lab it up again... but I dont believe refuse_t38 responded to > the T.38 invite at all. > > > > > > > On 3 May 2013 06:22, Michael Collins wrote: > >> >> >> >> On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: >> >>> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >>> >> >> Thanks for doing that. It appears that our T38 content on the wiki is >> woefully thin: >> http://wiki.freeswitch.org/wiki/T.38 >> >> If anyone out there has T.38 experience/knowledge/battle scars and they >> are willing to share that would be most appreciated. If you have specific >> configurations that you can post we could start with that. Also, if anyone >> has logs and/or SIP captures of T.38 exchanges and would be willing to post >> those please do so, or contact me off list and we'll talk about getting >> this item some more documentation. >> -MC >> >> >>> >>> I hope this helps... >>> >>> Jo?o Mesquita >>> FreeSWITCH? Solutions >>> >>> -MC >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/48300be6/attachment.html From jmesquita at freeswitch.org Fri May 3 07:20:05 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 3 May 2013 00:20:05 -0300 Subject: [Freeswitch-users] T38 Re-INVITE: FS responds 100 Trying, nothing further happens In-Reply-To: References: <44160CC4-1A0B-4AEC-A810-FFEAC3EB10C6@freeswitch.org> Message-ID: <88E79244-EC43-49C7-801F-A76D10812634@freeswitch.org> I hear ya. Tell you what, I'll lab some of the Jira tomorrow so I can help a bit. Sent from my iPhone On May 3, 2013, at 12:06 AM, jay binks wrote: > yea yea .. I know > > I already have enought jira's opened. > I cant keep up with providing the feedback to bugs I submit :P > > not enough hours in the day. > > > > > On 3 May 2013 12:14, Jo?o Mesquita wrote: >> If it didn't we would be happy to respond to the Jira opened! :) >> >> Sent from my iPhone >> >> On May 2, 2013, at 10:43 PM, jay binks wrote: >> >>> im 90% sure I tried that... >>> I trolled the source looking for all the hidden gems .. >>> >>> ill have to lab it up again... but I dont believe refuse_t38 responded to the T.38 invite at all. >>> >>> >>> >>> >>> >>> >>> On 3 May 2013 06:22, Michael Collins wrote: >>>> >>>> >>>> >>>> On Thu, May 2, 2013 at 11:48 AM, Jo?o Mesquita wrote: >>>>> if (switch_true(switch_channel_get_variable(channel, "refuse_t38"))) { >>>> >>>> Thanks for doing that. It appears that our T38 content on the wiki is woefully thin: >>>> http://wiki.freeswitch.org/wiki/T.38 >>>> >>>> If anyone out there has T.38 experience/knowledge/battle scars and they are willing to share that would be most appreciated. If you have specific configurations that you can post we could start with that. Also, if anyone has logs and/or SIP captures of T.38 exchanges and would be willing to post those please do so, or contact me off list and we'll talk about getting this item some more documentation. >>>> -MC >>>> >>>>> >>>>> I hope this helps... >>>>> >>>>> Jo?o Mesquita >>>>> FreeSWITCH? Solutions >>>> -MC >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > -- > Sincerely > > Jay > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/aeeac3dc/attachment-0001.html From smfarrukh at live.com Fri May 3 10:12:38 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Fri, 3 May 2013 06:12:38 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com>, , Message-ID: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster at davri.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130503/ed981dd3/attachment.html From 8f27e956 at gmail.com Fri May 3 11:02:43 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Fri, 3 May 2013 03:02:43 -0400 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: <-6910859898959998736@unknownmsgid> If you're sure it's not the end points, than is traffic is passing thru a stateful firewall (either direction/either side) where the SFW's UDP state timers may be too fast. /S ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-05-03, at 2:17, Farrukh Ali wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh ------------------------------ Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster at davri.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh ------------------------------ From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are: 1) Use session timers 2) There is a parameter to hang up a call is no RTP has been received for a certain period of time 3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.comFreeSWITCH-users mailing list FreeSWITCH-users at 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://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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.comFreeSWITCH-users mailing list FreeSWITCH-users at 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://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/20130503/78e5f1bd/attachment-0001.html From mehroz.ashraf85 at gmail.com Fri May 3 11:40:57 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 3 May 2013 00:40:57 -0700 (PDT) Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> <1367502588238-7590307.post@n2.nabble.com> <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> <1367507609182-7590309.post@n2.nabble.com> Message-ID: <1367566857945-7590327.post@n2.nabble.com> My configuration of SIP profile are: param name="nat-options-ping" value="true" param name="all-reg-options-ping" value="true" param name="unregister-on-options-fail" value="true" param name="enable-timer" value="false" and this is because, i was experiencing video session starting up within audio call, SDP session update once session timers completes, That was totally unexplained, and i eventually had to disable timer, which ultimately resolved that issue. What left is , RTP timers, and i have param name="rtp-timer-name" value="soft" param name="rtp-timeout-sec" value="15" but it is not helpful. Channel hangs up once only a client becomes unreachable. But if both becomes unreachable, Freeswitch is helpless!! which logs are required to dig it in details and what approach shall be considered? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590327.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Fri May 3 11:57:23 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 3 May 2013 08:57:23 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec -Steve On 3 May 2013 07:12, Farrukh Ali wrote: > Yes BDF, you are right, but I have been using Asterisk for a while and > never faced such problem, there has to be some mechanism Freeswitch is > lagging, beside this Freeswitch is quiet flexible to use due to its unique > XML configuration style, > And thanks for response > > Regards > Muhammad Farrukh > > ------------------------------ > Date: Thu, 2 May 2013 08:51:42 -0400 > From: bdfoster at davri.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Just a note: > > In your setup, a timeout if some sort is your only option. It may not be > efficient, it may not be perfect, but neither are your WiFi/Mobile clients > apparently. > > -BDF > On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: > > Thanks everyone, and Steve timers might not be efficient solution but for > now it will be good for testing purpose, could you please tell me exactly > which file and parameter should I look for to change, is this configuration > located in sofia.conf.xml ? kindly reply! > and thanks to all again! > > Regards, > Muhammad Farrukh > > ------------------------------ > From: steveayre at gmail.com > Date: Tue, 30 Apr 2013 17:20:36 +0100 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > SIP over UDP has no way of knowing the other side has lost connectivity > (eg wifi down). Sip droid will know the local network is down and be able > to hang up the call because of that. FreeSWITCH has no such knowledge, so > won't know the call is hung up and will continue sending RTP. > > 3 possibilities are: > 1) Use session timers > 2) There is a parameter to hang up a call is no RTP has been received for > a certain period of time > 3) Use SIP/TCP, keepalives might help (but the default is measured in > hours) > > Note that all of these are timeouts - you simply won't be able to know the > call has ended immediately. > > Steve > > > > On 30 Apr 2013, at 08:33, Farrukh Ali wrote: > > Hi, > > I need some help, there is an issue in Freeswitch with SIP clients, when a > call is not hung up properly Freeswitch does not close its RTP channels, > and it starts to get load on bandwidth after too many dead channels are > generated, Improper hang up means if during call sip client went out of the > range of WiFi, the client i.e SipDroid shows call hang up but freeswitch > continues to send voice packets. Kindly reply > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/8d512d8a/attachment.html From sertys at gmail.com Fri May 3 12:33:09 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 3 May 2013 10:33:09 +0200 Subject: [Freeswitch-users] ZRTP issue being hard to track Message-ID: I make currented the production fs a few weeks ago, because i've been unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup ). Trusted mitm works, but once i enable proxy-media the UAs ignore the zrtp-hash. One of them always says 'other side doesnt seem to support zrto'. I talked on irc to steevenielson to ask him any hints and he pointed the only difference between me and ostel.me setup is the rather old version he has. Could you elaborate with me on this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/e43976b5/attachment.html From sertys at gmail.com Fri May 3 12:36:25 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 3 May 2013 10:36:25 +0200 Subject: [Freeswitch-users] Client TLS certificate setup In-Reply-To: <1367529235228-7590319.post@n2.nabble.com> References: <1367529235228-7590319.post@n2.nabble.com> Message-ID: What kind of certificate are you using? Self -signed, chained with intermediate or root-verified? On May 3, 2013 12:30 AM, "adahary" wrote: > > I have already installed the CA certificate in the FS conf/ssl (following FS/TLS wiki) and managed to get TLS registering with PhonerLite so I know that basically it should work with other sip/tls clients as well. I'm trying to setup other softphones with TLS/SIPS connection to Freeswitch and I'm failing to register with CSimpleSip/android and Jitsi/windows. I get this: tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0xb6e02268): new connection from tls/1.2.3.4:13886/sips tport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATING tport_tls.c:873 tls_connect() tls_connect(0xb6e02268): events NEGOTIATING tport_tls.c:962 tls_connect() tls_connect(0xb6e02268): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2092 tport_close() tport_close(0xb6e02268): tls/1.2.3.4:13886/sips I will appriciate any help/instruction/tip on how to setup working TLS connection with CSimpleSip/android and Jitsi/windows clients. Thanks > ________________________________ > View this message in context: Client TLS certificate setup > 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://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/20130503/ac03ce67/attachment-0001.html From vermeulen.deon at gmail.com Fri May 3 12:54:41 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Fri, 03 May 2013 09:54:41 +0100 Subject: [Freeswitch-users] Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack Message-ID: <51837B51.5020305@gmail.com> Hi All Due to time constraints I'm posting here for the hope for a quick response. I'm currently trying to setup a Trunk to our upstream Carrier. The issues we are facing is that inbound calls from them are working 100%. Outbound calls to them are returned with a 500 error. The only immediate differences we could pickup was the SIP versions. They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 Is there any possibility someone could help me figure out what compatibility issues between the SIP versions could cause, especially for out bound calls to the Carrier? Thank you very much Kind Regards Deon Vermeulen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/adbea2aa/attachment.html From steveayre at gmail.com Fri May 3 13:13:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 3 May 2013 10:13:37 +0100 Subject: [Freeswitch-users] Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack In-Reply-To: <51837B51.5020305@gmail.com> References: <51837B51.5020305@gmail.com> Message-ID: 1.3.0.23 is their software version, not the SIP protocol version. Both will be using SIP v2. Some different devices do have different interpretations of that standard though, which can lead to interop problems between different device types. I suggest you contact your carrier to ask why they're sending 500 - they may be able to advise you what is causing this error. -Steve On 3 May 2013 09:54, Deon Vermeulen wrote: > Hi All > > Due to time constraints I'm posting here for the hope for a quick response. > > I'm currently trying to setup a Trunk to our upstream Carrier. > The issues we are facing is that inbound calls from them are working 100%. > Outbound calls to them are returned with a 500 error. > > The only immediate differences we could pickup was the SIP versions. > > They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. > I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 > > Is there any possibility someone could help me figure out what > compatibility issues between the SIP versions could cause, especially for > out bound calls to the Carrier? > > > Thank you very much > > > Kind Regards > Deon Vermeulen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/3f09f534/attachment.html From steveayre at gmail.com Fri May 3 13:16:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 3 May 2013 10:16:01 +0100 Subject: [Freeswitch-users] Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack In-Reply-To: References: <51837B51.5020305@gmail.com> Message-ID: Spotted this on the sofia configuration page: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Note: If your switch requires the timer option; for instance, Huawei > SoftX3000, it needs this optional field and drops the calls with "Session > Timer Check Message Failed", then you may be able to revert back the commit > that took away the Require: timer option which is an optional field by: Perhaps this is the issue? According to the wiki that issue requires a patch to the FreeSWITCH source, check the wiki link above. -Steve On 3 May 2013 10:13, Steven Ayre wrote: > 1.3.0.23 is their software version, not the SIP protocol version. Both > will be using SIP v2. > > Some different devices do have different interpretations of that standard > though, which can lead to interop problems between different device types. > > I suggest you contact your carrier to ask why they're sending 500 - they > may be able to advise you what is causing this error. > > -Steve > > > > On 3 May 2013 09:54, Deon Vermeulen wrote: > >> Hi All >> >> Due to time constraints I'm posting here for the hope for a quick >> response. >> >> I'm currently trying to setup a Trunk to our upstream Carrier. >> The issues we are facing is that inbound calls from them are working 100%. >> Outbound calls to them are returned with a 500 error. >> >> The only immediate differences we could pickup was the SIP versions. >> >> They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. >> I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 >> >> Is there any possibility someone could help me figure out what >> compatibility issues between the SIP versions could cause, especially for >> out bound calls to the Carrier? >> >> >> Thank you very much >> >> >> Kind Regards >> Deon Vermeulen >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130503/8cb84ea7/attachment.html From mbodbg at gmx.net Fri May 3 13:19:21 2013 From: mbodbg at gmx.net (mbo) Date: Fri, 3 May 2013 11:19:21 +0200 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode Message-ID: I'm referring to a two years old bug report http://jira.freeswitch.org/browse/FS-1309. Is it in the meantime somehow possible to map reply to SendMsg in asyc mode? I'm wondering why this ticket has been closed as "Won't fix", in my opinion is an essential feature to handle events properly. If not, I want to implement the work around described in the ticket, to set a channel variable in a round trip before executing the real command. Do I need to wait for the ChannelExecuteComplete event for the Set command, or can I send my "real" command right away after the Set command? Thanks Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/86b5f2b4/attachment.html From torstein.knutsen at gmail.com Fri May 3 13:44:16 2013 From: torstein.knutsen at gmail.com (Torstein Knutsen) Date: Fri, 3 May 2013 11:44:16 +0200 Subject: [Freeswitch-users] XML encoding attribute ? Message-ID: Hi I have a questione regarding XML output in event socket or cli. Example : "+" in the fs_cli when issuing "conference xml_list" returns *%2B*4726262626 When in fact my caller_id_name is "+4726262626". The printout shows no encoding attribute : I'm I correct to assume that if I can get encoding in utf-8, the "+" will be "+" and not "*%2B"* in the listing ? Does anybody know how I can accomplish this ? Im running ubuntu server 10.04 / FreeSWITCH Version 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 16:56:47Z) best regards Torstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/78b36988/attachment.html From eagle.antonio at gmail.com Fri May 3 13:49:58 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 03 May 2013 10:49:58 +0100 Subject: [Freeswitch-users] XML encoding attribute ? In-Reply-To: References: Message-ID: <51838846.90801@gmail.com> FS Events Are _URL Encoded_ So *%2B = "+" * http://www.degraeve.com/reference/urlencoding.php *Regards A/t * On 5/3/13 10:44 AM, Torstein Knutsen wrote: > Hi > > I have a questione regarding XML output in event socket or cli. > Example : "+" in the fs_cli when issuing "conference xml_list" > returns *%2B*4726262626 > When in fact my caller_id_name is "+4726262626". > > > The printout shows no encoding attribute : > > > I'm I correct to assume that if I can get encoding in utf-8, the "+" > will be "+" and not "*%2B"* in the listing ? > > Does anybody know how I can accomplish this ? > Im running ubuntu server 10.04 / FreeSWITCH Version > 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 16:56:47Z) > > best regards > Torstein > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/c7e9c527/attachment-0001.html From torstein.knutsen at gmail.com Fri May 3 14:13:44 2013 From: torstein.knutsen at gmail.com (Torstein Knutsen) Date: Fri, 3 May 2013 12:13:44 +0200 Subject: [Freeswitch-users] XML encoding attribute ? In-Reply-To: <51838846.90801@gmail.com> References: <51838846.90801@gmail.com> Message-ID: Hi Thanks for your spport. So that means that there is no way around "conference xml_list" displaying %2B in my googling around for this I found many printouts where the encoding attribute in the XML was present, I was wondering if that had something to do with it (as it's missing in my installation) .. br Torstein On 3 May 2013 11:49, Antonio Teixeira wrote: > FS Events Are *URL Encoded* So > > *%2B = "+" > * > http://www.degraeve.com/reference/urlencoding.php > > *Regards > A/t > * > > On 5/3/13 10:44 AM, Torstein Knutsen wrote: > > Hi > > I have a questione regarding XML output in event socket or cli. Example > : "+" in the fs_cli when issuing "conference xml_list" returns > *%2B*4726262626 > When in fact my caller_id_name is "+4726262626". > > > The printout shows no encoding attribute : > > > I'm I correct to assume that if I can get encoding in utf-8, the "+" > will be "+" and not "*%2B"* in the listing ? > > Does anybody know how I can accomplish this ? > Im running ubuntu server 10.04 / FreeSWITCH Version > 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 16:56:47Z) > > best regards > Torstein > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130503/dd39366d/attachment.html From eagle.antonio at gmail.com Fri May 3 14:25:14 2013 From: eagle.antonio at gmail.com (Antonio Teixeira) Date: Fri, 03 May 2013 11:25:14 +0100 Subject: [Freeswitch-users] XML encoding attribute ? In-Reply-To: References: <51838846.90801@gmail.com> Message-ID: <5183908A.4080703@gmail.com> Sincerely i don't know BUT you are probably using esl to make that request. You could use your native language functions. Like for PHP http://php.net/manual/en/function.urldecode.php OR Use the FS APi http://wiki.freeswitch.org/wiki/Mod_commands#url_decode Workflow : eslData = esl.Query(conference xml_list) eslCleanData = url.decode(eslData) ..... A/t:) On 5/3/13 11:13 AM, Torstein Knutsen wrote: > Hi > > Thanks for your spport. > > So that means that there is no way around "conference xml_list" > displaying %2B > in my googling around for this I found many printouts where the > encoding attribute in the XML was present, I was wondering if that had > something to do with it (as it's missing in my installation) .. > > br > Torstein > > > > > > On 3 May 2013 11:49, Antonio Teixeira > wrote: > > FS Events Are _URL Encoded_ So > > *%2B = "+" > * > http://www.degraeve.com/reference/urlencoding.php > > *Regards > A/t > * > > On 5/3/13 10:44 AM, Torstein Knutsen wrote: >> Hi >> >> I have a questione regarding XML output in event socket or cli. >> Example : "+" in the fs_cli when issuing "conference xml_list" >> returns *%2B*4726262626 >> When in fact my caller_id_name is "+4726262626". >> >> >> The printout shows no encoding attribute : >> >> >> I'm I correct to assume that if I can get encoding in utf-8, the >> "+" will be "+" and not "*%2B"* in the listing ? >> >> Does anybody know how I can accomplish this ? >> Im running ubuntu server 10.04 / FreeSWITCH Version >> 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 >> 16:56:47Z) >> >> best regards >> Torstein >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/326252b9/attachment.html From julien.terrasson at gmail.com Fri May 3 14:49:15 2013 From: julien.terrasson at gmail.com (julien terrasson) Date: Fri, 3 May 2013 12:49:15 +0200 Subject: [Freeswitch-users] How to retreive FreeSWITCH-IPv4 value ? Message-ID: Hello, I'm trying to dump the Freeswitch-IPv4 value in one of the session. But for some reason the getvar primitive returns an "_undef_" value Should i use a different primitive to retreive that value ? Example : freeswitch at debian> uuid_dump *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* Event-Name: CHANNEL_DATA Core-UUID: 73c3dc70-b3d4-11e2-b885-f9b7dfbfcfcb FreeSWITCH-Hostname: debian FreeSWITCH-Switchname: debian *FreeSWITCH-IPv4: 192.168.1.2* FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2013-05-03%2012%3A28%3A45 .... freeswitch at debian> uuid_getvar *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* * FreeSWITCH-IPv4* *_undef_* Regards, Julien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/4d3fbc8b/attachment.html From m.hubert at hexanet.fr Fri May 3 14:56:36 2013 From: m.hubert at hexanet.fr (Mickael Hubert) Date: Fri, 3 May 2013 12:56:36 +0200 Subject: [Freeswitch-users] multiple same header in INVITE Message-ID: Hi list, I have my freeswitch in SBC "mode" and topology hiding. I'd like to copy some header between A leg and B leg (not to copy all headers). My code is OK, but when I have multiple same headers in my INVITE, freeswitch only copy the last header. Ex: A Leg: Diversion: ; reason=user-busy; privacy=off Diversion: ; reason=user-busy; privacy=full ;counter=3 B leg: Diversion: ;reason=user-busy;privacy=full;counter=3 Only last Diversion header is copied. how can I copy all same header please ? my code: ** * * * * ** thanks in advance -- Cordialement Hubert Micka?l Ing?nieur VOIP - Hexanet -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/23489b64/attachment-0001.html From ira at connectmevoice.com Fri May 3 15:22:23 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Fri, 3 May 2013 07:22:23 -0400 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <20130430230019.60e244cc@mail.tritonwest.net> References: <20130430230019.60e244cc@mail.tritonwest.net> Message-ID: I downloaded a clean clone. If I build the ManagedESL solution first (using Release,x64). It tells me esl.lib is missed. So, I assume that I have to build the Freeswitch.2010 solution first (release,x64). So I build that, then the ManagedESL solution. My application creates a ESLConnection fine, I subscribe to events fine, but when I get an event, I get a memory corruption exception. When I use the dll's that you posted, everything works great. Also, when you initially load the ManagedESL solution, so you always get a yellow "!" on the test projects app.config? It seems to be missing. I just would love to figure this out. Its driving me crazy :). Thanks. -- Ira Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com On Tue, Apr 30, 2013 at 7:00 PM, Dave R. Kompel wrote: > ** > I don't remember each time I have to figure it out. However the one thing > that I do remember is that you can only build it from a clean clone, if you > have built FS there first, the build will fail, cause output files conflict. > > --Dave > > ------------------------------ > *From:* Ira Tessler [mailto:ira at connectmevoice.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Tue, 30 Apr 2013 14:12:57 -0700 > > *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers > (Windows) > > When I am building the esl.dll and ManagedEsl.dll, first I load the > Freeswitch 2010 solution, select the build configuration of Release, x64 > and build the solution. Then I load the ManagedEsl solution and so the > same. Is this correct? > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com > > > On Mon, Apr 29, 2013 at 11:29 AM, Gregor Nanger wrote: > >> Great Dave! >> >> Thank you. >> >> I also had hard time compiling DLLs and also have memory corruption.. Do >> not remember exactly, but it has to be right combination of build >> configuration... >> >> Gregor >> >> >> 2013/4/27 Dave R. Kompel >> >>> ** >>> Same thing happened to me. It seems that for some reason if I build it >>> on the same machine, and from the same clone that I build the rest of FS I >>> get the same results. So every time there is a major change to the ESL >>> stuff, I build from fresh clone on a different machine, where nothing else >>> in the tree has been built, verfy size of the native DLL, zip, and push up >>> there, since I use it in a number of projects. >>> >>> If I can ever figure out the NuGet powershell magic to build a script >>> into a NuGet package that will add the build step to also copy the native >>> interop dll as well as the managed one, I'll push a HuGet package up with >>> it, and have a CI build trigger on changes to the ESL library. >>> >>> --Dave >>> >>> ------------------------------ >>> *From:* Ira Tessler [mailto:ira at connectmevoice.com] >>> *To:* FreeSWITCH Users Help [mailto: >>> freeswitch-users at lists.freeswitch.org] >>> *Sent:* Sat, 27 Apr 2013 05:32:41 -0700 >>> >>> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >>> (Windows) >>> >>> Ok Thanks for the binaries. They work great! I complied them myself. I >>> set the build configuration to release and x64. When my app tries to access >>> the esl.dll, I get a memory corruption (i don't remember the exact >>> exception) exception. It drives me crazy! :) >>> >>> Ira Tessler >>> Lead Software Engineer >>> ConnectMe >>> (732) 490-9007 x2 >>> ira at connectmevoice.com >>> >>> >>> On Fri, Apr 26, 2013 at 12:44 PM, Dave R. Kompel wrote: >>> >>>> ** >>>> Ira, >>>> >>>> It is a pain, since you have to have our dev box set up right. For that >>>> reason I keep this: http://download.drknetworking.com/eslmanaged.zip around >>>> for ppl, it has both native wrappers in it. >>>> >>>> --Dave >>>> >>>> ------------------------------ >>>> *From:* Ira Tessler [mailto:ira at connectmevoice.com] >>>> *To:* FreeSWITCH Users Help [mailto: >>>> freeswitch-users at lists.freeswitch.org] >>>> *Sent:* Fri, 26 Apr 2013 04:18:22 -0700 >>>> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >>>> (Windows) >>>> >>>> I am having the same trouble building the ESL. Would you be able to >>>> post the steps you did to build both the 32 and 64 bit versions is esl.dll >>>> and managedesl.dll? >>>> >>>> Thanks for the binaries! >>>> >>>> --Ira >>>> >>>> Ira Tessler >>>> Lead Software Engineer >>>> ConnectMe >>>> (732) 490-9007 x2 >>>> ira at connectmevoice.com >>>> >>>> >>>> On Mon, Feb 4, 2013 at 8:25 PM, Dave R. Kompel wrote: >>>> >>>>> ** >>>>> Getting it to swig right, and getting the native DLL built for X64 was >>>>> give me a problem, till I did it on a diferent box. >>>>> >>>>> I notice you added the VS2012 projects for it as well about an hour >>>>> ago :) Thanks. >>>>> >>>>> --Dave >>>>> >>>>> ------------------------------ >>>>> *From:* Jeff Lenk [mailto:jeff at jefflenk.com] >>>>> *To:* freeswitch-users at lists.freeswitch.org >>>>> *Sent:* Mon, 04 Feb 2013 16:37:20 -0800 >>>>> *Subject:* Re: [Freeswitch-users] ESL Client Library Managed wrappers >>>>> (Windows) >>>>> >>>>> Dave I'm curious what was wrong? The solution should build without >>>>> changing >>>>> anything. >>>>> >>>>> You have to build FreeSWITCH for the platform you want then open the >>>>> esl >>>>> solution and select the platform you want. The only caveat here is that >>>>> because you are building code that is platform specific that you have >>>>> to >>>>> specify the correct platform and not "Any Cpu" or "Mixed Platform". >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.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://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 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/b73e6901/attachment-0001.html From adahary at gmail.com Fri May 3 13:30:51 2013 From: adahary at gmail.com (adahary) Date: Fri, 3 May 2013 02:30:51 -0700 (PDT) Subject: [Freeswitch-users] Client TLS certificate setup In-Reply-To: References: <1367529235228-7590319.post@n2.nabble.com> Message-ID: <1367573451256-7590335.post@n2.nabble.com> I've followed:http://wiki.freeswitch.org/wiki/SIP_TLSagent.pem => ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org.cafile.pem => ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org.client.pem => ./gentls_cert create_client ...also generated pkcs12 which was installed successfuly is both jitsi and csimplesip (but fail to register):openssl pkcs12 -export -out certificate.pfx -inkey privateKey.key -in certificate.crt -certfile CACert.crtThis is the TLS part of vars.xml: The reload mod_sofia log shows that it is loading the .pem files from the conf/ssl.any tls registration tls/1.2.3.4/sips from jitsi/csimplesip get rejected with certificate error.thanks for your support -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-tp7590319p7590335.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/2fa84ec5/attachment.html From navnath.sonavne at yahoo.com Fri May 3 16:20:22 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Fri, 3 May 2013 20:20:22 +0800 (SGT) Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 10 In-Reply-To: References: Message-ID: <1367583622.77006.YahooMailNeo@web192203.mail.sg3.yahoo.com> Hi, As you said create new extensions for flex client and h323 endpoint(Ekiga softphone). I created two new extensions 1103.xml for flex client and 1104.xml for ekiga soft phone. I login using 1103 on flex client and make call to 1104. Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga phone rings and when i answer the? call ,call terminates immediately by showing local user rejected call. In another case where I call from Ekiga phone(1104) to flex client(1103),it says person at 1104 is? not available and says record your voice mail.In this case flex client did not get any incoming call. Here is my default.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Correct me if my dialstrings are wrong. Tell me if any other changes to made in other files also. How can I make both way calling ?Please help me its emergency , I have demo next week.? -Navnath. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, 2 May 2013 8:59 PM Subject: FreeSWITCH-users Digest, Vol 83, Issue 10 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. make ekiga to flex call (Navnath Sonavne) ? 2. Re: make ekiga to flex call (Brian Foster) ? 3. Re: Check if UA is still there? (mehroz) ? 4. Re: Check if UA is still there? (Ken Rice) ? 5. Re: Check if UA is still there? (mehroz) ? 6. Re: Check if UA is still there? (Michael Collins) Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source? to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension.? After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes? in default.xml accordingly to forward calls to this ip. Here is My default.xml part : ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? Ekiga phone can answer the call and there is audio?transmission?from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to?flex?client. How to call flex client from ekiga softphone? Anybody?please?me on this issue. -Navnath. According to your dialplan you're forcing all calls that hit Local_Extension (any call with a destination number of 4 digits) to call your ekiga softphone. Your other problem lies in the fact that you should have different dialplans for different methods of calling unlike endpoints one for h232 one for flex. See below: -BDF On May 2, 2013 9:05 AM, "Navnath Sonavne" wrote: > > Hi All, > > I am using flex client(at 192.168.9.165) given in freeswitch source? > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). > I have two users registered on freeswitch with extension as 1100 and 1101 in default context. > I loged in using one of extension in flex client,then I dail another extension.? > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes? > in default.xml accordingly to forward calls to this ip. > Here is My default.xml part : > > > ? ? ? > ? ? ? > ? ? ? > ? ? ? < ? ? ? > ? ? ? > ? ? ? > ? ? ? < ? ? ? ? ?? > > > Ekiga phone can answer the call and there is audio?transmission?from both end successfully. > > But this is one way calling i.e. from flex to ekiga. > Now I want to call from Ekiga softphone to?flex?client. > How to call flex client from ekiga softphone? > > Anybody?please?me on this issue. > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > What if we want to even hungup the channels associated with those users? does freeswitch helps ? Like, i have a scenario in which , an active call disrupts , once BOTH users gets disconnected from network. Here, both users gets unregistered after option messages but the call sustains. Is there any thing i can handle this scenario? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html Sent from the freeswitch-users mailing list archive at Nabble.com. options pings are really just for registered users. a user does not have to be registered to place a call. more useful in your scenario is rtp timers and session timers Ken Sent from my iPad On May 2, 2013, at 8:49, mehroz wrote: > What if we want to even hungup the channels associated with those users? does > freeswitch helps ? > > Like, i have a scenario in which , an active call disrupts , once BOTH users > gets disconnected from network. > Here, both users gets unregistered after option messages but the call > sustains. Is there any thing i can handle this scenario? > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.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://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 i cannot use session-timers, disabled due to an issue with video calling.... and rtp timers just does not seems to be working! i have But once network disconnects from both clients (works good, with a single client network outage), FS does not hangs up the call ..... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html Sent from the freeswitch-users mailing list archive at Nabble.com. On Thu, May 2, 2013 at 8:13 AM, mehroz wrote: i cannot use session-timers, disabled due to an issue with video calling.... > >and rtp timers just does not seems to be working! > Can you elaborate on this? Do you have a full pcap and FS debug log of a call with RTP in both directions and then the incoming RTP stream stopping and yet the call not being disconnected? >i have > >But once network disconnects from both clients (works good, with a single >client network outage), FS does not hangs up the call ..... > Correct. No SIP signaling and no session timers means there's nothing left to tell FS that the call should be torn down, except for the RTP timers mentioned above. That's why you'll need to supply a complete debug w/ pcap in order to find out why RTP timers are not working. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20130503/bf79e08f/attachment-0001.html From tnsampaio at bsd.com.br Fri May 3 16:54:24 2013 From: tnsampaio at bsd.com.br (Tiago Sampaio) Date: Fri, 03 May 2013 09:54:24 -0300 Subject: [Freeswitch-users] Blind transfer with call waiting In-Reply-To: References: <5182C1C4.40804@bsd.com.br> Message-ID: <5183B380.2000503@bsd.com.br> Its exactly what i need! Thx. But i supose it may work with att_xfer, correct? Here when i hangup when start att_xfer call hungup... Em 02-05-2013 17:12, Anthony Minessale escreveu: > you could try > http://wiki.freeswitch.org/wiki/Variable_campon > > > > On Thu, May 2, 2013 at 2:43 PM, Tiago Sampaio > wrote: > > Is there something like blind transfer with call waiting when the > target > is busy? > > I have a situation here, EX: When Jhon (user 1001) answer a call > and and > customer ask to talk to Robert (user 1002), > Jhon do a blind tansfer an hangup. Currently im wasting that call when > Robert is busy. I need an way to park the > call and when Robert become available ring his extension (and > maybe put > an timeout and call back to Jhon). > > Im currently writing an script with perl to do this, something > like that > (i do this from memory): > > use strict; > use POSIX qw(strftime); > my ($target_ext,$source_ext,$tryes); > my $max_time = 30; > > our $session; > > $target_ext = $session->getVariable('target_ext'); > $source_ext = $session->getVariable('source_ext'); > > while($tryes <= $max_time){ > my $uuid = $session->execute("db","select/in_a_call/$target_ext"); > > if(!$uuid) > $session->execute("trasfer","$target_ext XML > transfer_context"); > > sleep(1); > $tryes++; > } > > $session->execute("transfer","$source_ext XML transfer_context"); > > > Before call an extension i do db insert/in_a_call//${uuid} > and after db delete/in_a_call/ > > But in thinking will be too hard to maintain this code and must be an > easier way to do this... > > Any help? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/c53851e8/attachment.html From adahary at gmail.com Fri May 3 17:10:58 2013 From: adahary at gmail.com (adahary) Date: Fri, 3 May 2013 06:10:58 -0700 (PDT) Subject: [Freeswitch-users] Client TLS certificate setup In-Reply-To: References: <1367529235228-7590319.post@n2.nabble.com> Message-ID: <1367586658270-7590345.post@n2.nabble.com> Daniel,Now I see that I get the 'err 26:unsupported certificate purpose' for the fail reason.I have checked the purpose and found 'TLS Web..' - should be ok.What could be the reason?fs_client log:tport.c:2745 tport_wakeup_pri() tport_wakeup_pri(0x9a97ea0): events INtport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x9a97ea0): new secondary tport 0x9b03ea0tport_type_tls.c:607 tport_tls_accept() tport_tls_accept(0x9b03ea0): new connection from tls/62.90.161.235:50438/sipstport_tls.c:873 tls_connect() tls_connect(0x9b03ea0): events NEGOTIATINGtport_tls.c:873 tls_connect() tls_connect(0x9b03ea0): events NEGOTIATINGtport_tls.c:253 tls_verify_cb() -Error with certificate at depth: 0tport_tls.c:255 tls_verify_cb() issuer = /CN=il1.mobi2save.com/O=mobi2save.comtport_tls.c:257 tls_verify_cb() subject = /CN=il1.mobi2save.com/O=mobi2save.comtport_tls.c:258 tls_verify_cb() *err 26:unsupported certificate purpose*tport_tls.c:962 tls_connect() tls_connect(0x9b03ea0): TLS setup failed (error:00000001:lib(0):func(0):reason(1))tport.c:2092 tport_close() tport_close(0x9b03ea0): tls/62.90.161.235:50438/sipsfreeswitch at 127.0.0.1:8028 at internal>[root at il1 ssl]# openssl x509 -in client.pem -text -nooutCertificate: Data: Version: 3 (0x2) Serial Number: b2:68:02:6b:19:d3:aa:36 Signature Algorithm: sha1WithRSAEncryption Issuer: CN=il1.mobi2save.com, O=mobi2save.com Validity Not Before: May 2 19:20:09 2013 GMT Not After : May 1 19:20:09 2019 GMT Subject: CN=il1.mobi2save.com, O=mobi2save.com Subject Public Key Info: Public Key Algorithm: rsaEncryption RSA Public Key: (2048 bit) Modulus (2048 bit): 00:cf:81:c9:62:5a:0b:d0:0e:2e:5b:7b:21:bf:9e: b9:50:3a:bc:91:5b:93:21:8c:87:8d:f2:1b:df:24: 19:7a:4a:0d:e3:39:00:7f:a8:5d:d3:8f:c6:67:90: 60:cb:53:ee:c9:74:b0:74:d9:fe:90:7d:15:bf:82: 3d:89:cb:49:6a:54:96:65:72:01:d8:12:a8:23:63: 85:bd:a6:e4:c6:12:86:45:d3:8f:c2:ea:58:34:b5: 0e:a5:89:b5:fe:d6:8f:f3:9e:cb:2b:cc:5e:f3:b1: ff:30:d2:b6:8f:c0:af:70:a7:bc:2c:c6:1d:79:3a: bc:87:07:5e:70:ca:d9:9c:c7:91:d5:25:47:92:62: 55:47:df:c6:0b:38:55:a5:c1:d1:e3:98:47:5f:be: 90:84:05:41:6f:84:1e:4c:7b:0d:d4:21:6f:20:12: f5:d9:73:0e:bf:0c:31:df:86:40:86:56:91:f5:dc: 6d:30:32:8b:b1:9c:09:82:b7:f4:ec:18:1e:7b:9f: 41:a1:49:84:3f:01:a9:ea:d5:0b:37:81:a5:3c:58: af:31:92:b4:db:53:9f:6b:05:08:7b:34:d1:62:9f: 23:54:4a:c2:2b:eb:c0:9a:c3:9d:da:ae:72:19:24: 1c:5f:62:68:01:b9:0f:5e:9e:04:7a:5b:6d:ce:06: 03:c1 Exponent: 65537 (0x10001) X509v3 extensions: Netscape Comment: FS Client Cert X509v3 Basic Constraints: CA:FALSE X509v3 Subject Key Identifier: 33:41:5C:37:CF:8B:B3:C6:45:72:28:81:6A:97:FB:7D:D4:EF:41:AE X509v3 Authority Key Identifier: DirName:/CN=il1.mobi2save.com/O=mobi2save.com serial:B4:B8:71:80:AC:28:33:48 X509v3 Subject Alternative Name: DNS:il1.mobi2save.com Netscape Cert Type: SSL Client X509v3 Extended Key Usage: TLS Web Client Authentication Signature Algorithm: sha1WithRSAEncryption 5f:46:da:81:89:6f:2e:60:9f:f8:fb:8c:a9:87:d1:53:7f:78: b4:0c:98:ab:fc:93:53:41:4f:24:24:71:02:1e:59:92:ca:08: 47:f4:3f:2f:da:3f:f0:d8:4c:5b:69:24:d1:29:f7:9d:d7:95: 0d:a0:25:5d:4a:6e:04:69:c4:4e:58:77:ba:24:11:59:14:7d: 23:4c:e3:c3:27:df:8e:cc:c0:30:1e:29:c3:94:c3:a6:05:23: 76:60:0a:aa:6e:7d:a0:fc:12:c8:49:96:41:b9:1f:3c:8c:d8: 8a:fa:a3:14:5b:11:67:26:6d:85:57:2d:10:86:fa:65:62:12: e9:8b:6a:a8:2b:dc:0c:70:3e:3d:f6:2d:97:9a:82:41:5f:99: fe:67:f7:7c:f3:48:4e:2a:2d:d0:32:46:77:a4:00:05:3d:be: 26:4d:d9:92:9b:92:8e:78:ac:01:5b:a0:29:fa:9c:69:c1:74: 86:26:ce:e3:fa:b3:40:b5:59:bb:b3:fe:27:91:4a:4f:2b:89: 0e:bd:e6:7a:ca:28:8f:64:31:71:5b:77:4d:65:2a:77:30:7d: 69:21:0c:54:77:6e:2e:8c:d2:72:35:ad:8f:e7:f0:04:34:cb: da:25:40:ec:14:9b:34:dd:60:ad:0a:39:d9:df:91:11:66:9c: 03:ee:4a:d7 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Client-TLS-certificate-setup-tp7590319p7590345.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/6458c064/attachment.html From anthony.minessale at gmail.com Fri May 3 18:39:14 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 3 May 2013 09:39:14 -0500 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" wrote: > If you're on VoIP and an endpoint completely disappears (eg wifi down) > then the is absolutely no way for the other endpoint to detect that. That > goes whatever software you're using. > > It's impossible for the down endpoint to tell the other that it's down, > since it's down. Lack of messages also isn't enough to know it's down - > it's normal during most of the SIP call for there to be no SIP packets, and > RTP is designed to cope with packet loss. > > Absolutely the only way to detect it is to spot when you haven't heard > from them for a while and timeout. Using SIP/UDP that's either in > signalling with session timers (send a SIP packet, wait for a reply, > timeout if no response within N seconds), or in media with RTP timers > (hangup call if no RTP received for N seconds). > > To use session timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout > > Or to use RTP timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec > > -Steve > > > > > On 3 May 2013 07:12, Farrukh Ali wrote: > >> Yes BDF, you are right, but I have been using Asterisk for a while and >> never faced such problem, there has to be some mechanism Freeswitch is >> lagging, beside this Freeswitch is quiet flexible to use due to its unique >> XML configuration style, >> And thanks for response >> >> Regards >> Muhammad Farrukh >> >> ------------------------------ >> Date: Thu, 2 May 2013 08:51:42 -0400 >> From: bdfoster at davri.com >> >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Dead channels in freeswitch >> >> Just a note: >> >> In your setup, a timeout if some sort is your only option. It may not be >> efficient, it may not be perfect, but neither are your WiFi/Mobile clients >> apparently. >> >> -BDF >> On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: >> >> Thanks everyone, and Steve timers might not be efficient solution but for >> now it will be good for testing purpose, could you please tell me exactly >> which file and parameter should I look for to change, is this configuration >> located in sofia.conf.xml ? kindly reply! >> and thanks to all again! >> >> Regards, >> Muhammad Farrukh >> >> ------------------------------ >> From: steveayre at gmail.com >> Date: Tue, 30 Apr 2013 17:20:36 +0100 >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Dead channels in freeswitch >> >> SIP over UDP has no way of knowing the other side has lost connectivity >> (eg wifi down). Sip droid will know the local network is down and be able >> to hang up the call because of that. FreeSWITCH has no such knowledge, so >> won't know the call is hung up and will continue sending RTP. >> >> 3 possibilities are: >> 1) Use session timers >> 2) There is a parameter to hang up a call is no RTP has been received for >> a certain period of time >> 3) Use SIP/TCP, keepalives might help (but the default is measured in >> hours) >> >> Note that all of these are timeouts - you simply won't be able to know >> the call has ended immediately. >> >> Steve >> >> >> >> On 30 Apr 2013, at 08:33, Farrukh Ali wrote: >> >> Hi, >> >> I need some help, there is an issue in Freeswitch with SIP clients, when >> a call is not hung up properly Freeswitch does not close its RTP channels, >> and it starts to get load on bandwidth after too many dead channels are >> generated, Improper hang up means if during call sip client went out of the >> range of WiFi, the client i.e SipDroid shows call hang up but freeswitch >> continues to send voice packets. Kindly reply >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server 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-usersUNSUBSCRIBE: >> 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://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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The >> CudaTel Communication Server 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-usersUNSUBSCRIBE: >> 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://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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130503/2da03cbb/attachment-0001.html From vermeulen.deon at gmail.com Fri May 3 19:01:05 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Fri, 03 May 2013 16:01:05 +0100 Subject: [Freeswitch-users] Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack In-Reply-To: References: <51837B51.5020305@gmail.com> Message-ID: <5183D131.3020709@gmail.com> Hi Steven I just want to give an update. The problem is resolved without having to do any patches or funny configurations. Some Technician other then the 2 I was working with sent me the error info and I immediately figured what it was. **0.105506817 SE2300-SBC0 SIP/8/debug:Sip get intercom signal query address by ip[192.168.0.250-->10.222.0.200] route failed.* The problems was that I was listening on port 6090 and they on 5060. With the invite I send them the return port was not allowed on their SBC. I changed my port to 5060 and calls started flowing. I guess the timer issue could have been resolved in the latest version? Thanks again for helping. Kind Regards Deon > Steven Ayre > May 3, 2013 10:16 AM > Spotted this on the sofia configuration page: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout > > > Perhaps this is the issue? According to the wiki that issue requires a > patch to the FreeSWITCH source, check the wiki link above. > > -Steve > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > Steven Ayre > May 3, 2013 10:13 AM > 1.3.0.23 is their software version, not the SIP protocol version. Both > will be using SIP v2. > > Some different devices do have different interpretations of that > standard though, which can lead to interop problems between different > device types. > > I suggest you contact your carrier to ask why they're sending 500 - > they may be able to advise you what is causing this error. > > -Steve > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > Deon Vermeulen > May 3, 2013 9:54 AM > Hi All > > Due to time constraints I'm posting here for the hope for a quick > response. > > I'm currently trying to setup a Trunk to our upstream Carrier. > The issues we are facing is that inbound calls from them are working 100%. > Outbound calls to them are returned with a 500 error. > > The only immediate differences we could pickup was the SIP versions. > > They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. > I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 > > Is there any possibility someone could help me figure out what > compatibility issues between the SIP versions could cause, especially > for out bound calls to the Carrier? > > > Thank you very much > > > Kind Regards > Deon Vermeulen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/bcee1ff1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: compose-unknown-contact.jpg Type: image/jpeg Size: 770 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/bcee1ff1/attachment.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: postbox-contact.jpg Type: image/jpeg Size: 1143 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/bcee1ff1/attachment-0001.jpg From msc at freeswitch.org Fri May 3 21:22:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 May 2013 10:22:09 -0700 Subject: [Freeswitch-users] Check if UA is still there? In-Reply-To: <1367566857945-7590327.post@n2.nabble.com> References: <4DEE7DD5.4000900@telefaks.de> <4DEE8E3F.6050107@telefaks.de> <1367502588238-7590307.post@n2.nabble.com> <23A88D80-1D6C-443A-80F1-9FDFCCA2EF06@freeswitch.org> <1367507609182-7590309.post@n2.nabble.com> <1367566857945-7590327.post@n2.nabble.com> Message-ID: On Fri, May 3, 2013 at 12:40 AM, mehroz wrote: > My configuration of SIP profile are: > > param name="nat-options-ping" value="true" > param name="all-reg-options-ping" value="true" > param name="unregister-on-options-fail" value="true" > > param name="enable-timer" value="false" > and this is because, i was experiencing video session starting up within > audio call, SDP session update once session timers completes, That was > totally unexplained, and i eventually had to disable timer, which > ultimately > resolved that issue. > > > What left is , RTP timers, and i have > param name="rtp-timer-name" value="soft" > param name="rtp-timeout-sec" value="15" > but it is not helpful. Channel hangs up once only a client becomes > unreachable. But if both becomes unreachable, Freeswitch is helpless!! > > > which logs are required to dig it in details and what approach shall be > considered? > Same thing I mentioned before: pcap with both SIP and RTP debug log of call not behaving properly. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/88ad2e30/attachment.html From msc at freeswitch.org Fri May 3 21:27:10 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 May 2013 10:27:10 -0700 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: On Fri, May 3, 2013 at 2:19 AM, mbo wrote: > I'm referring to a two years old bug report > http://jira.freeswitch.org/browse/FS-1309. > > Is it in the meantime somehow possible to map reply to SendMsg in asyc > mode? I'm wondering why this ticket has been closed as "Won't fix", in my > opinion is an essential feature to handle events properly. > > If not, I want to implement the work around described in the ticket, to > set a channel variable in a round trip before executing the real command. > Do I need to wait for the ChannelExecuteComplete event for the Set command, > or can I send my "real" command right away after the Set command? > To be safe you should verify that the set app actually ran before you implement the workaround. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/fa4dfaae/attachment-0001.html From william.king at quentustech.com Fri May 3 21:39:23 2013 From: william.king at quentustech.com (William King) Date: Fri, 03 May 2013 12:39:23 -0500 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <1360024640390-7587003.post@n2.nabble.com> References: <20130203204955.8db66002@mail.tritonwest.net> <1360024640390-7587003.post@n2.nabble.com> Message-ID: <5183F64B.9090602@quentustech.com> Can someone update the expected instructions on how to build the Windows ESL client on the wiki? http://wiki.freeswitch.org/wiki/Event_Socket#.NET_Client_library 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 02/04/2013 06:37 PM, Jeff Lenk wrote: > Dave I'm curious what was wrong? The solution should build without changing > anything. > > You have to build FreeSWITCH for the platform you want then open the esl > solution and select the platform you want. The only caveat here is that > because you are building code that is platform specific that you have to > specify the correct platform and not "Any Cpu" or "Mixed Platform". > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7587003.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://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 From jleung at v10networks.ca Fri May 3 22:39:48 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Fri, 3 May 2013 11:39:48 -0700 Subject: [Freeswitch-users] Friday Free For All Message-ID: <00c501ce482d$984acc10$c8e06430$@v10networks.ca> It's Friday and you know what we do today ;) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/1dd5416d/attachment.html From msc at freeswitch.org Fri May 3 23:01:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 May 2013 12:01:40 -0700 Subject: [Freeswitch-users] How to retreive FreeSWITCH-IPv4 value ? In-Reply-To: References: Message-ID: Julien, "FreeSWITCH-IPv4" is not a channel variable. Rather, it is an event header. Even so, I'm not sure that an individual event is necessarily the best place from which to be extracting this value. What problem are you trying to solve? Also, how many IP addresses do you have on your system? -MC On Fri, May 3, 2013 at 3:49 AM, julien terrasson wrote: > Hello, > > I'm trying to dump the Freeswitch-IPv4 value in one of the session. > But for some reason the getvar primitive returns an "_undef_" value > Should i use a different primitive to retreive that value ? > > Example : > > freeswitch at debian> uuid_dump *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* > > Event-Name: CHANNEL_DATA > Core-UUID: 73c3dc70-b3d4-11e2-b885-f9b7dfbfcfcb > FreeSWITCH-Hostname: debian > FreeSWITCH-Switchname: debian > *FreeSWITCH-IPv4: 192.168.1.2* > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2013-05-03%2012%3A28%3A45 > .... > > freeswitch at debian> uuid_getvar *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* * > FreeSWITCH-IPv4* > > *_undef_* > > Regards, > > Julien > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/b1a9f737/attachment.html From msc at freeswitch.org Fri May 3 23:10:15 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 May 2013 12:10:15 -0700 Subject: [Freeswitch-users] Blind transfer with call waiting In-Reply-To: <5183B380.2000503@bsd.com.br> References: <5182C1C4.40804@bsd.com.br> <5183B380.2000503@bsd.com.br> Message-ID: I've never tried campon with the att_xfer app. Try it out and let us know if it works! -MC On Fri, May 3, 2013 at 5:54 AM, Tiago Sampaio wrote: > Its exactly what i need! Thx. > > But i supose it may work with att_xfer, correct? > Here when i hangup when start att_xfer call hungup... > > > Em 02-05-2013 17:12, Anthony Minessale escreveu: > > you could try > http://wiki.freeswitch.org/wiki/Variable_campon > > > > On Thu, May 2, 2013 at 2:43 PM, Tiago Sampaio wrote: > >> Is there something like blind transfer with call waiting when the target >> is busy? >> >> I have a situation here, EX: When Jhon (user 1001) answer a call and and >> customer ask to talk to Robert (user 1002), >> Jhon do a blind tansfer an hangup. Currently im wasting that call when >> Robert is busy. I need an way to park the >> call and when Robert become available ring his extension (and maybe put >> an timeout and call back to Jhon). >> >> Im currently writing an script with perl to do this, something like that >> (i do this from memory): >> >> use strict; >> use POSIX qw(strftime); >> my ($target_ext,$source_ext,$tryes); >> my $max_time = 30; >> >> our $session; >> >> $target_ext = $session->getVariable('target_ext'); >> $source_ext = $session->getVariable('source_ext'); >> >> while($tryes <= $max_time){ >> my $uuid = $session->execute("db","select/in_a_call/$target_ext"); >> >> if(!$uuid) >> $session->execute("trasfer","$target_ext XML transfer_context"); >> >> sleep(1); >> $tryes++; >> } >> >> $session->execute("transfer","$source_ext XML transfer_context"); >> >> >> Before call an extension i do db insert/in_a_call//${uuid} >> and after db delete/in_a_call/ >> >> But in thinking will be too hard to maintain this code and must be an >> easier way to do this... >> >> Any help? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/346fdd75/attachment-0001.html From anthony.minessale at gmail.com Fri May 3 23:13:01 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 3 May 2013 14:13:01 -0500 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: Compromise: Update to latest and sub to the "private_command" event, you should get back you own event with channel data merged in. On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: > > > On Fri, May 3, 2013 at 2:19 AM, mbo wrote: > >> I'm referring to a two years old bug report >> http://jira.freeswitch.org/browse/FS-1309. >> >> Is it in the meantime somehow possible to map reply to SendMsg in asyc >> mode? I'm wondering why this ticket has been closed as "Won't fix", in my >> opinion is an essential feature to handle events properly. >> >> If not, I want to implement the work around described in the ticket, to >> set a channel variable in a round trip before executing the real command. >> Do I need to wait for the ChannelExecuteComplete event for the Set command, >> or can I send my "real" command right away after the Set command? >> > To be safe you should verify that the set app actually ran before you > implement the workaround. > -MC > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/69bcb31a/attachment.html From msc at freeswitch.org Fri May 3 23:14:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 May 2013 12:14:30 -0700 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 10 In-Reply-To: <1367583622.77006.YahooMailNeo@web192203.mail.sg3.yahoo.com> References: <1367583622.77006.YahooMailNeo@web192203.mail.sg3.yahoo.com> Message-ID: Put a console debug log on pastebin.freeswitch.org so that we can see what's going on. Also, see this page for some handy troubleshooting tips: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin -MC On Fri, May 3, 2013 at 5:20 AM, Navnath Sonavne wrote: > Hi, > > As you said create new extensions for flex client and h323 endpoint(Ekiga > softphone). > I created two new extensions 1103.xml for flex client and 1104.xml for > ekiga soft phone. > I login using 1103 on flex client and make call to 1104. > Now when I call from flex client(1103) to Ekiga softphone(1104),ekiga > phone rings and when i answer the > call ,call terminates immediately by showing local user rejected call. > > In another case where I call from Ekiga phone(1104) to flex > client(1103),it says person at 1104 is > not available and says record your voice mail.In this case flex client did > not get any incoming call. > > > Here is my default.xml > > > > > data="effective_caller_id_number=${dialed_extension}"/> > > > > > > > > > data="effective_caller_id_number=${dialed_extension}"/> > > > > > > > > Correct me if my dialstrings are wrong. > Tell me if any other changes to made in other files also. > How can I make both way calling ?Please help me its emergency , I have > demo next week. > > -Navnath. > > > > > > ------------------------------ > *From:* "freeswitch-users-request at lists.freeswitch.org" < > freeswitch-users-request at lists.freeswitch.org> > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Thursday, 2 May 2013 8:59 PM > *Subject:* FreeSWITCH-users Digest, Vol 83, Issue 10 > > ----- Forwarded Message ----- > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. make ekiga to flex call (Navnath Sonavne) > 2. Re: make ekiga to flex call (Brian Foster) > 3. Re: Check if UA is still there? (mehroz) > 4. Re: Check if UA is still there? (Ken Rice) > 5. Re: Check if UA is still there? (mehroz) > 6. Re: Check if UA is still there? (Michael Collins) > Hi All, > > I am using flex client(at 192.168.9.165) given in freeswitch source > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via > freeswitch server(at 192.168.8.41). > I have two users registered on freeswitch with extension as 1100 and 1101 > in default context. > I loged in using one of extension in flex client,then I dail another > extension. > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have > made changes > in default.xml accordingly to forward calls to this ip. > Here is My default.xml part : > > > > > data="effective_caller_id_number=${dialed_extension}"/> > > > > > > > > > Ekiga phone can answer the call and there is audio transmission from both > end successfully. > > But this is one way calling i.e. from flex to ekiga. > Now I want to call from Ekiga softphone to flex client. > How to call flex client from ekiga softphone? > > Anybody please me on this issue. > > -Navnath. > > > > > According to your dialplan you're forcing all calls that hit > Local_Extension (any call with a destination number of 4 digits) to call > your ekiga softphone. Your other problem lies in the fact that you should > have different dialplans for different methods of calling unlike endpoints > one for h232 one for flex. > See below: > -BDF > On May 2, 2013 9:05 AM, "Navnath Sonavne" > wrote: > > > > Hi All, > > > > I am using flex client(at 192.168.9.165) given in freeswitch source > > to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via > freeswitch server(at 192.168.8.41). > > I have two users registered on freeswitch with extension as 1100 and > 1101 in default context. > > I loged in using one of extension in flex client,then I dail another > extension. > > After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have > made changes > > in default.xml accordingly to forward calls to this ip. > > Here is My default.xml part : > > > > > > > > > > data="effective_caller_id_number=${dialed_extension}"/> > > < should never hardcode a specific endpoint IP if this dialplan should be > dialing others as well > > > > > > > > < you set your dialplan to bridge the call to your softphone > > > > > > > > Ekiga phone can answer the call and there is audio transmission from > both end successfully. > > > > But this is one way calling i.e. from flex to ekiga. > > Now I want to call from Ekiga softphone to flex client. > > How to call flex client from ekiga softphone? > > > > Anybody please me on this issue. > > > > -Navnath. > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > What if we want to even hungup the channels associated with those users? > does > freeswitch helps ? > > Like, i have a scenario in which , an active call disrupts , once BOTH > users > gets disconnected from network. > Here, both users gets unregistered after option messages but the call > sustains. Is there any thing i can handle this scenario? > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > options pings are really just for registered users. a user does not have > to be registered to place a call. more useful in your scenario is rtp > timers and session timers > > Ken > Sent from my iPad > > On May 2, 2013, at 8:49, mehroz wrote: > > > What if we want to even hungup the channels associated with those users? > does > > freeswitch helps ? > > > > Like, i have a scenario in which , an active call disrupts , once BOTH > users > > gets disconnected from network. > > Here, both users gets unregistered after option messages but the call > > sustains. Is there any thing i can handle this scenario? > > > > > > > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590307.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://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 > > > i cannot use session-timers, disabled due to an issue with video > calling.... > > and rtp timers just does not seems to be working! > > i have > > But once network disconnects from both clients (works good, with a single > client network outage), FS does not hangs up the call ..... > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-tp6451091p7590309.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > > On Thu, May 2, 2013 at 8:13 AM, mehroz wrote: > > i cannot use session-timers, disabled due to an issue with video > calling.... > > and rtp timers just does not seems to be working! > > Can you elaborate on this? Do you have a full pcap and FS debug log of a > call with RTP in both directions and then the incoming RTP stream stopping > and yet the call not being disconnected? > > > i have > > But once network disconnects from both clients (works good, with a single > client network outage), FS does not hangs up the call ..... > > Correct. No SIP signaling and no session timers means there's nothing left > to tell FS that the call should be torn down, except for the RTP timers > mentioned above. That's why you'll need to supply a complete debug w/ pcap > in order to find out why RTP timers are not working. > > -MC > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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://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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/760b5df1/attachment-0001.html From miha at softnet.si Fri May 3 23:39:54 2013 From: miha at softnet.si (Miha) Date: Fri, 03 May 2013 21:39:54 +0200 Subject: [Freeswitch-users] Alarm, KISS OFF In-Reply-To: <517F1003.1070207@coppice.org> References: <517A7C63.8080204@softnet.si> <517A7EF2.8000005@softnet.si> <517A837F.7020800@softnet.si> <1366984077.4742.257.camel@luna.madrid.commsmundi.com> <517E21F0.2020905@coppice.org> <517E27A0.5050000@softnet.si> <517F1003.1070207@coppice.org> Message-ID: Steve, so there is nothing I can do? It must be something on line? Is there any function/application that I can set so that this will help alarm device? Thanks! Miha On Tue, 30 Apr 2013 08:27:47 +0800 Steve Underwood wrote: > The commonest alarm protocol is Ademco Contact ID and it > variants. This > is very undemanding of the channel, and it would take a > pretty broken > VoIP connection to stop them working. VoIP channels tend > to fail for > alarms when people ignore the fact that VoIP usually uses > mains powered > terminals, with no battery backup. A simple power outage > means no > 911/999/112/whatever calls, and no alarm protocols. > That's the real > weakness. > > Steve > > On 04/30/2013 02:15 AM, Steven Schoch wrote: > > Speaking of alarm systems over VoIP, here's a story of > an alarm > > company that was sued by the estate of a murdered woman > because her > > alarm system failed to notify the central station when > the attacker > > broke in. It turns out she had switched to VoIP a few > weeks earlier, > > which caused problems: > > > http://www.security.honeywell.com/hsc/documents/alarmorg_VoIP.pdf > > > > Here's something from an alarm company: > > http://www.thealarmcompany.com/files/voip%281%29.pdf > > And from another alarm company: > > > http://bentleyalarm.com/why-voip-is-bad-for-your-alarm-system/ > > > > Bottom line: I'd think really hard before switching an > alarm system to > > VoIP. When we switched the company to VoIP, I was able > to get a very > > inexpensive measured line for our alarm system. (We > also use it for > > incoming FAX, because it works.) > > > > -- > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > 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 From nneul at mst.edu Sat May 4 00:33:28 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 03 May 2013 15:33:28 -0500 Subject: [Freeswitch-users] Voicemail greeting length? Message-ID: <51841F18.9000500@mst.edu> Seeing a problem where recording a voicemail greeting is truncated on freeswitch. The strange thing is - FS acts like it's recording the entire duration I'm speaking (30-40 seconds), but when you play back the recording, it stops after 16-17 seconds fairly consistently. (Same thing if I tell it to save and then go and look at the actual wav file.) There's no indication of anything happening in the logs. I'm only seeing the truncated recording with calls to VM from skinny phones. It does not affect voicemail message recording at all, only greetings. This is with current trunk. Any suggestions on places to start looking? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From nneul at mst.edu Sat May 4 01:04:17 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 03 May 2013 16:04:17 -0500 Subject: [Freeswitch-users] Voicemail greeting length? In-Reply-To: <51841F18.9000500@mst.edu> References: <51841F18.9000500@mst.edu> Message-ID: <51842651.20008@mst.edu> Setting record_waste_resources to true or low or high values had no effect. -- Nathan On 05/03/2013 03:33 PM, Nathan Neulinger wrote: > Seeing a problem where recording a voicemail greeting is truncated on freeswitch. The strange thing is - FS acts like > it's recording the entire duration I'm speaking (30-40 seconds), but when you play back the recording, it stops after > 16-17 seconds fairly consistently. (Same thing if I tell it to save and then go and look at the actual wav file.) > There's no indication of anything happening in the logs. > > I'm only seeing the truncated recording with calls to VM from skinny phones. > > It does not affect voicemail message recording at all, only greetings. > > This is with current trunk. > > > > Any suggestions on places to start looking? -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From royj at yandex.ru Sat May 4 01:15:59 2013 From: royj at yandex.ru (royj) Date: Sat, 4 May 2013 01:15:59 +0400 Subject: [Freeswitch-users] ZRTP issue being hard to track In-Reply-To: References: Message-ID: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> May be in profile would change picture On Fri, 3 May 2013 10:33:09 +0200 Daniel Ivanov wrote: > I make currented the production fs a few weeks ago, because i've been > unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup ). > Trusted mitm works, but once i enable proxy-media the UAs ignore the > zrtp-hash. One of them always says 'other side doesnt seem to support > zrto'. I talked on irc to steevenielson to ask him any hints and he pointed > the only difference between me and ostel.me setup is the rather old version > he has. Could you elaborate with me on this? -- royj From richardrcruzc at gmail.com Sat May 4 01:26:58 2013 From: richardrcruzc at gmail.com (Richard Cruz) Date: Fri, 3 May 2013 17:26:58 -0400 Subject: [Freeswitch-users] settings for a Trixbox v 2.6 connection to a Sip Trunking translate to gateway using Freeswitch 1.2.0 Message-ID: Hi There! I have the following settings for a Trixbox v 2.6 connection to a Sip Trunking at tricom.net Host=XXX.X.132.37 defaultuser=********** Secret= *********** Fromuser=xxxxxxxxxx Type=friend insecure=invite,port Dtmfmode=rfc2833 Disallow=all Allow=g729 Canredirect=no Canreivite=no Context=from-trunk Qualify=yes and I want to translate to gateway using Freeswitch 1.2.0 -- Richard Cruz 678.394-6400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130503/c0eecf75/attachment.html From julien.terrasson at gmail.com Sat May 4 11:55:31 2013 From: julien.terrasson at gmail.com (julien terrasson) Date: Sat, 4 May 2013 09:55:31 +0200 Subject: [Freeswitch-users] How to retreive FreeSWITCH-IPv4 value ? (Julien Terrasson) Message-ID: Hi Michael, I'm actually using a single IP address. But i'm trying to think of an architecture where multiple freeswitch would be in use, each of them acting as user-controled recorders. The reason i wanted to have the Freeswitch-IPv4 from the dialplan was to be able to include it when defining the record name : Ex : -Record__.wav That way i can get the information of where the recording took place. But in the end i used a workaround : i defined a server_id var in the vars.xml.. But anyway, thanks for the prompt feedback ! Have a nice week-end. Julien ------------------------------------------------------------------------------------------------------------------------------------ Julien, "FreeSWITCH-IPv4" is not a channel variable. Rather, it is an event header. Even so, I'm not sure that an individual event is necessarily the best place from which to be extracting this value. What problem are you trying to solve? Also, how many IP addresses do you have on your system? -MC From jeff at jefflenk.com Sat May 4 21:38:35 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Sat, 4 May 2013 10:38:35 -0700 (PDT) Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <5183F64B.9090602@quentustech.com> References: <20130203204955.8db66002@mail.tritonwest.net> <1360024640390-7587003.post@n2.nabble.com> <5183F64B.9090602@quentustech.com> Message-ID: <1367689115874-7590362.post@n2.nabble.com> I made some small additions to the Wiki for this but I don't really understand why there are so many problems building the code. I also committed some changes to head which should simplify the solution for those building the code. I just built the x64 debug solution code and it works without issue. I am happy to improve the process if any problems are identified with patches provided to Jira. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7590362.html Sent from the freeswitch-users mailing list archive at Nabble.com. From drk at drkngs.net Sat May 4 23:41:46 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sat, 04 May 2013 12:41:46 -0700 Subject: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) In-Reply-To: <1367689115874-7590362.post@n2.nabble.com> Message-ID: <20130504194146.b24b7cb1@mail.tritonwest.net> Because ppl's machines arn't set up the same as your development machine. Including you have stuff like swig in c:\dev... --Dave _____ From: Jeff Lenk [mailto:jeff at jefflenk.com] To: freeswitch-users at lists.freeswitch.org Sent: Sat, 04 May 2013 10:38:35 -0700 Subject: Re: [Freeswitch-users] ESL Client Library Managed wrappers (Windows) I made some small additions to the Wiki for this but I don't really understand why there are so many problems building the code. I also committed some changes to head which should simplify the solution for those building the code. I just built the x64 debug solution code and it works without issue. I am happy to improve the process if any problems are identified with patches provided to Jira. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ESL-Client-Library-Managed-wrappers-Windows-tp7586973p7590362.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://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/20130504/9057323b/attachment.html From sdevoy at bizfocused.com Sat May 4 23:51:49 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 4 May 2013 15:51:49 -0400 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones Message-ID: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> HI, Can anyone recommend a good quality WiFi (wireless IP) VOIP Phone that works with FS? Looking for < $200 and top quality functionality. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130504/e83bd299/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130504/e83bd299/attachment-0001.gif From spencer at 5ninesolutions.com Sun May 5 00:17:59 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Sat, 4 May 2013 15:17:59 -0500 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> Message-ID: <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic KX-TGP500s are great. -----Original message----- From: Sean Devoy To: FreeSWITCH-users at lists.freeswitch.org Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130504/7d84c918/attachment.html From bpriddy at bryantschools.org Sun May 5 00:24:56 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Sat, 4 May 2013 15:24:56 -0500 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Message-ID: The SNOM DECT model is inexpensive and really good as well On May 4, 2013 3:23 PM, "Spencer Thomason" wrote: > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great. > > > -----Original message----- > > *From: *Sean Devoy * > To: *FreeSWITCH-users at lists.freeswitch.org* > Sent: *Sat, May 4, 2013 19:56:55 GMT+00:00* > Subject: *[Freeswitch-users] WiFi IP VOIP SIP phones > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130504/ce8822ce/attachment.html From max at nysolutions.com Sun May 5 06:42:45 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Sun, 5 May 2013 02:42:45 +0000 Subject: [Freeswitch-users] spam>spam> WiFi IP VOIP SIP phones In-Reply-To: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> Message-ID: The Unidata WPU_7800. Also have a look at the Spectralink though it is more expensive. You can also use an android device that has sip or an app. The many Snom and Grandstream desk phones can have Wi-Fi added with a Wi-Fi dongle. 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 Sean Devoy Sent: Saturday, May 04, 2013 3:52 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: spam>spam>[Freeswitch-users] WiFi IP VOIP SIP phones HI, Can anyone recommend a good quality WiFi (wireless IP) VOIP Phone that works with FS? Looking for < $200 and top quality functionality. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... 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Name: image003.jpg Type: image/jpeg Size: 2424 bytes Desc: image003.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130505/768bfd52/attachment.jpg From luis.daniel.lucio at gmail.com Sun May 5 06:44:06 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 4 May 2013 22:44:06 -0400 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Message-ID: I use at home snom m9 without problems El may 4, 2013 4:28 p.m., "Blake Priddy" escribi?: > The SNOM DECT model is inexpensive and really good as well > On May 4, 2013 3:23 PM, "Spencer Thomason" > wrote: > >> Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic >> KX-TGP500s are great. >> >> >> -----Original message----- >> >> *From: *Sean Devoy * >> To: *FreeSWITCH-users at lists.freeswitch.org* >> Sent: *Sat, May 4, 2013 19:56:55 GMT+00:00* >> Subject: *[Freeswitch-users] WiFi IP VOIP SIP phones >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130504/a447118b/attachment-0001.html From nathandownes at hotmail.com Sun May 5 07:16:02 2013 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sun, 5 May 2013 13:16:02 +1000 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Message-ID: I?ve used the Snom m3 and m9.. the Siemens c610ip and a510 are way better in feel/call quality and cheaper, not tried any others yet From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: Sunday, 5 May 2013 12:44 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones I use at home snom m9 without problems El may 4, 2013 4:28 p.m., "Blake Priddy" escribi?: The SNOM DECT model is inexpensive and really good as well On May 4, 2013 3:23 PM, "Spencer Thomason" wrote: Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic KX-TGP500s are great. -----Original message----- From: Sean Devoy To: FreeSWITCH-users at lists.freeswitch.org Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130505/819e2f52/attachment.html From sertys at gmail.com Sun May 5 15:49:58 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 5 May 2013 13:49:58 +0200 Subject: [Freeswitch-users] ZRTP issue being hard to track In-Reply-To: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> References: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> Message-ID: I have missed this param, cannot wait to try on monday. Will report if that's it. On May 4, 2013 12:19 AM, "royj" wrote: > May be in profile would > change picture > > On Fri, 3 May 2013 10:33:09 +0200 > Daniel Ivanov wrote: > > > I make currented the production fs a few weeks ago, because i've been > > unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup > ). > > Trusted mitm works, but once i enable proxy-media the UAs ignore the > > zrtp-hash. One of them always says 'other side doesnt seem to support > > zrto'. I talked on irc to steevenielson to ask him any hints and he > pointed > > the only difference between me and ostel.me setup is the rather old > version > > he has. Could you elaborate with me on this? > > > -- > royj > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/537824dc/attachment.html From joelrosenfield at yahoo.com Sat May 4 21:55:51 2013 From: joelrosenfield at yahoo.com (Joel Rosenfield) Date: Sat, 4 May 2013 10:55:51 -0700 (PDT) Subject: [Freeswitch-users] Is it possible to record a conference with only one attendee? Message-ID: <1367690151.38623.YahooMailNeo@web162203.mail.bf1.yahoo.com> I am starting/stopping the conference recording for an existing conference using the api commands "conference recording start/stop ". I have didn't find a conference flag that would cause the audio to be recorded when there is just one person in the conference. ?Did I miss something? I did see that the auto-record flag only starts recording when there are two attendees, and I wonder if what I am observing when trying to record with only one attendee happens for a similar reason. I did notice that when I played an announcement on a 1-person conference, that announcement does get recorded: ?conference play My scenario for when I start a recording is this with just myself on the conference is this: (1) play an annoucement (2) I say something (3) wait for someone to join and say I some more. In my recording file, I hear the announcement (1), then silence, then I hear the audio from the two-person conference (3). ?I do not hear my own voice (2). The use case is that when people try the recording feature, the first thing they want to do is try it out themselves before they adopt it. Thanks, - Joel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130504/8640cd0b/attachment-0001.html From ash at url.net.au Sun May 5 04:38:42 2013 From: ash at url.net.au (Ashley Breeden) Date: Sun, 5 May 2013 10:38:42 +1000 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> Message-ID: Hi Sean, I have only come across one wifi based phone which was called a Escene WS320, but I probably would recommend it, it seems to work fine on a local network but not when your SIP server is externally hosted and the phone is going via NAT. I typically use the following when someone wants wifi: If it's a desk phone they need: - Use a Netgear WNCE2001 bridge with any ethernet based phone, personally we use Yealink for the desk phone. In Australia the WNCE2001 costs around $50 - http://www.netgear.com.au/landing/wnce2001.aspx If it's a full cordless solution you need: - Siemens Gigaset is good, DECT back to the base - http://gigaset.com/au/en/cms/PageInternetVoIPNextGPhones.html - Yealink have also just release an alternative to the Siemens phone called the W52P, initial tests seem to be good but I have only played with this model for about 30 mins - http://www.yealink.com/product_info.aspx?ProductsCateID=308&CateId=307&BaseInfoCateId=308&Cate_Id=308&parentcateid=307 Cheers, Ash. On 05/05/2013, at 5:51 AM, Sean Devoy wrote: > HI, > > Can anyone recommend a good quality WiFi (wireless IP) VOIP Phone that works with FS? > > Looking for < $200 and top quality functionality. > > Thanks, > Sean > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130505/e3215700/attachment-0001.html From lconroy at insensate.co.uk Sun May 5 17:05:09 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sun, 5 May 2013 14:05:09 +0100 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Message-ID: <128A8D2C-4B10-4503-A218-7D53C5E003BD@insensate.co.uk> Hi Folks, +1 -- I use multiple Siemens S675IP DECT 'phones. Voice quality good (supports G722 and speakerphone is OK, unlike some), range is excellent, Battery life good, and they just work (assuming you don't drop them out of a pocket into the loo, which is how most of my company's DECT 'phones die). BTW, note that, for large campus sites, there's always something like Their CAL pricing for what is a basic SIP server is laughable, but the DECT base stations and cellular controller are not too painful (i.e. it's a good fit with fS ;). I have to add; SNOM DECT 'phones are OK-good, Panys you either like or don't, but Polycom DECT phones are too toxic for landfill, IMHO. Problem I've seen with WIFi phones is battery life is not good, range is less than DECT, and I haven't seen a *really* new design for quite a while. Power is always going to be a problem, as WiFi Tx power is higher than DECT, and the link layer's more "chatty" on standby. all the best, Lawrence On 5 May 2013, at 04:16, Mr Nathan Downes wrote: > I?ve used the Snom m3 and m9.. the Siemens c610ip and a510 are way better in > feel/call quality and cheaper, not tried any others yet > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis > Daniel Lucio Quiroz > Sent: Sunday, 5 May 2013 12:44 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones > > > > I use at home snom m9 without problems > > El may 4, 2013 4:28 p.m., "Blake Priddy" > escribi?: > > The SNOM DECT model is inexpensive and really good as well > > On May 4, 2013 3:23 PM, "Spencer Thomason" > wrote: > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great. > > > > -----Original message----- > > From: Sean Devoy > To: FreeSWITCH-users at lists.freeswitch.org > Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 > Subject: [Freeswitch-users] WiFi IP VOIP SIP phones > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From luis.daniel.lucio at gmail.com Sun May 5 18:36:40 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sun, 5 May 2013 10:36:40 -0400 Subject: [Freeswitch-users] Billing software Message-ID: What other options for FS compatible software other than vBilling do you recommend me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130505/0b159d9e/attachment.html From vermeulen.deon at gmail.com Sun May 5 19:52:26 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Sun, 5 May 2013 16:52:26 +0100 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: Check out ASTPP Kind Regards Deon Vermeulen Sent from my iPhone On May 5, 2013, at 15:36, Luis Daniel Lucio Quiroz wrote: > What other options for FS compatible software other than vBilling do you recommend me? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ahmed at netelsat.net Sun May 5 19:56:42 2013 From: ahmed at netelsat.net (Ahmed Sboor) Date: Sun, 5 May 2013 20:56:42 +0500 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: Jerasoft's VCS , check http://www.jerasoft.net On Sun, May 5, 2013 at 7:36 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > What other options for FS compatible software other than vBilling do you > recommend me? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/a77d6c9c/attachment.html From sdevoy at bizfocused.com Sun May 5 20:10:28 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 5 May 2013 12:10:28 -0400 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> Message-ID: <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> Spencer and Blake, I have seen DECT SIP phones and been fairly confused. Is that wired Ethernet to the base and then a standard wireless home phone? My customer wants these for rooms that have no wired Ethernet and need 6 distinct (not shared) extensions. I don't think you can have 6 DECT phones (to unique bases) in one house can you? Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Blake Priddy Sent: Saturday, May 04, 2013 4:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones The SNOM DECT model is inexpensive and really good as well On May 4, 2013 3:23 PM, "Spencer Thomason" wrote: Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic KX-TGP500s are great. -----Original message----- From: Sean Devoy To: FreeSWITCH-users at lists.freeswitch.org Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130505/eb6139e5/attachment.html From bpriddy at bryantschools.org Sun May 5 20:17:53 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Sun, 5 May 2013 11:17:53 -0500 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> Message-ID: The Snom does have an Ethernet base. And you can have up to 10 to one station if I remember correctly. I will have to read on it again but tge operate on a 5ghz channel to the base station. They can all have different extensions. On May 5, 2013 11:14 AM, "Sean Devoy" wrote: > Spencer and Blake,**** > > I have seen DECT SIP phones and been fairly confused. Is that wired > Ethernet to the base and then a standard wireless home phone? My customer > wants these for rooms that have no wired Ethernet and need 6 distinct (not > shared) extensions. I don?t think you can have 6 DECT phones (to unique > bases) in one house can you?**** > > ** ** > > Thanks,**** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Blake Priddy > *Sent:* Saturday, May 04, 2013 4:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] WiFi IP VOIP SIP phones**** > > ** ** > > The SNOM DECT model is inexpensive and really good as well **** > > On May 4, 2013 3:23 PM, "Spencer Thomason" > wrote:**** > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great.**** > > > > -----Original message-----**** > > *From: *Sean Devoy * > To: *FreeSWITCH-users at lists.freeswitch.org* > Sent: *Sat, May 4, 2013 19:56:55 GMT+00:00* > Subject: *[Freeswitch-users] WiFi IP VOIP SIP phones**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/5208ca1f/attachment-0001.html From sdevoy at bizfocused.com Sun May 5 20:31:07 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 5 May 2013 12:31:07 -0400 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <128A8D2C-4B10-4503-A218-7D53C5E003BD@insensate.co.uk> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <128A8D2C-4B10-4503-A218-7D53C5E003BD@insensate.co.uk> Message-ID: <4d9901ce49ad$f1f8fe40$d5eafac0$@bizfocused.com> Thanks to everyone who answered. I am dealing with a language barrier with this customer so bear with me. Now that I understand where these phones are actually going, I can refine my query. The basic concept is to change a boarding house with 12 rooms on a single analog line to VOIP with individual extensions. 6 of the rooms are wired or will be added easily. The other 6 rooms cannot be wire but will have WiFi access. It sounds like WiFi desk phones would be fine. Single Line/Extension is good. So, can 6 WiFi/DECT phones on distinct bases coexist in the same radio range? Now I see Blake has already replied and the SNOM can have a single base with up to 10 non-shared extensions! I guess that would be a strategically placed base and 6 charger/handsets. I shall ready up on SNOM. Would WiFi dongles on Desk phones work better for this venue? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence Conroy Sent: Sunday, May 05, 2013 9:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones Hi Folks, +1 -- I use multiple Siemens S675IP DECT 'phones. Voice quality good (supports G722 and speakerphone is OK, unlike some), range is excellent, Battery life good, and they just work (assuming you don't drop them out of a pocket into the loo, which is how most of my company's DECT 'phones die). BTW, note that, for large campus sites, there's always something like Their CAL pricing for what is a basic SIP server is laughable, but the DECT base stations and cellular controller are not too painful (i.e. it's a good fit with fS ;). I have to add; SNOM DECT 'phones are OK-good, Panys you either like or don't, but Polycom DECT phones are too toxic for landfill, IMHO. Problem I've seen with WIFi phones is battery life is not good, range is less than DECT, and I haven't seen a *really* new design for quite a while. Power is always going to be a problem, as WiFi Tx power is higher than DECT, and the link layer's more "chatty" on standby. all the best, Lawrence On 5 May 2013, at 04:16, Mr Nathan Downes wrote: > I?ve used the Snom m3 and m9.. the Siemens c610ip and a510 are way > better in feel/call quality and cheaper, not tried any others yet > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Luis Daniel Lucio Quiroz > Sent: Sunday, 5 May 2013 12:44 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones > > > > I use at home snom m9 without problems > > El may 4, 2013 4:28 p.m., "Blake Priddy" > escribi?: > > The SNOM DECT model is inexpensive and really good as well > > On May 4, 2013 3:23 PM, "Spencer Thomason" > > wrote: > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great. > > > > -----Original message----- > > From: Sean Devoy > To: FreeSWITCH-users at lists.freeswitch.org > Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 > Subject: [Freeswitch-users] WiFi IP VOIP SIP phones > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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-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://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-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://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-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://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 From bdfoster at davri.com Sun May 5 20:42:04 2013 From: bdfoster at davri.com (Brian Foster) Date: Sun, 5 May 2013 12:42:04 -0400 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: JBilling, it's complex but extremely flexible. They have an open-source and commercial product. http://jbilling.com On May 5, 2013 12:01 PM, "Ahmed Sboor" wrote: > Jerasoft's VCS , check http://www.jerasoft.net > > On Sun, May 5, 2013 at 7:36 PM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> What other options for FS compatible software other than vBilling do you >> recommend me? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/5525ef1d/attachment.html From sertys at gmail.com Sun May 5 21:00:31 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 5 May 2013 19:00:31 +0200 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: Im both for and against astpp. It's relatively well designed, but code is black magic and hard to modify. It's been working well after i managed to get it runnin. On May 5, 2013 7:45 PM, "Brian Foster" wrote: > JBilling, it's complex but extremely flexible. They have an open-source > and commercial product. http://jbilling.com > On May 5, 2013 12:01 PM, "Ahmed Sboor" wrote: > >> Jerasoft's VCS , check http://www.jerasoft.net >> >> On Sun, May 5, 2013 at 7:36 PM, Luis Daniel Lucio Quiroz < >> luis.daniel.lucio at gmail.com> wrote: >> >>> What other options for FS compatible software other than vBilling do you >>> recommend me? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/a196737f/attachment.html From bdfoster at davri.com Sun May 5 21:10:15 2013 From: bdfoster at davri.com (Brian Foster) Date: Sun, 5 May 2013 13:10:15 -0400 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <4d9901ce49ad$f1f8fe40$d5eafac0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <128A8D2C-4B10-4503-A218-7D53C5E003BD@insensate.co.uk> <4d9901ce49ad$f1f8fe40$d5eafac0$@bizfocused.com> Message-ID: I've dealt with a similar situation. I work for a property management company, and at one stage we were developing a strategy to basically resell voip service to tenants. Each would have a phone and prepay for minutes. We even tbought about having a kiosk to top up their account when needed by credit card. They could also top up by paying a little extra on top of their rent which would then be applied to their voip account. We ended up not doing it. And here's why: 1. WiFi/DECT phones are too expensive to replace. They would get beat up or lost. It's just too risky. On top of that, WiFi just isn't great at all for real-time media such as voip. You could do some tweaking to mitigate some of the issues, but it's still not worth it. Use DECT if you really want to go down this route. You could rent the phones, but due to our clientele at this property, it wasn't feasible. 2. We thought about using an FXS channel bank and running copper to every room. It takes care of the lost phone issue, and the tenant would be responsible for their phone because, well, they bought it. If the tenant wouldn't have been able to afford it, we could have set them up on a payment plan for a cordless phone. In total, we would have needed 16 channels, plus one for the office. Basically, a 24 port channel bank. They are relatively inexpensive, especially considering the use of wifi/dect phones. Unfortunately, there is not much in it if you want any sort of ROI. It would take a few years to get it, and by that time you wouldn't be far from replacing the equipment you originally installed. That's also being optimistic, as that estimate would assume 70% occupancy. Now there are several ways we could have pursued this further, but then the Obamaphone came out and by then, there was no demand. So, I'm not saying it will never work. It's just difficult and requires some major number crunching to make sure you stay above water. If there were 50 units, that would be a major help. But this property only has 16. If you need some help, I'd be happy to, ping me off list. I spent six months off and on studying the feasibility of such a venture. There's a lot of things you need to be aware of, and you need to know your client's habits. -BDF Sent from my Samsung Galaxy S4 On May 5, 2013 12:36 PM, "Sean Devoy" wrote: > Thanks to everyone who answered. I am dealing with a language barrier with > this customer so bear with me. > > Now that I understand where these phones are actually going, I can refine > my > query. > > The basic concept is to change a boarding house with 12 rooms on a single > analog line to VOIP with individual extensions. 6 of the rooms are wired > or > will be added easily. The other 6 rooms cannot be wire but will have WiFi > access. It sounds like WiFi desk phones would be fine. Single > Line/Extension is good. > > So, can 6 WiFi/DECT phones on distinct bases coexist in the same radio > range? > Now I see Blake has already replied and the SNOM can have a single base > with > up to 10 non-shared extensions! I guess that would be a strategically > placed base and 6 charger/handsets. I shall ready up on SNOM. > > Would WiFi dongles on Desk phones work better for this venue? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Lawrence > Conroy > Sent: Sunday, May 05, 2013 9:05 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones > > Hi Folks, > +1 -- I use multiple Siemens S675IP DECT 'phones. > Voice quality good (supports G722 and speakerphone is OK, unlike some), > range is excellent, Battery life good, and they just work (assuming you > don't drop them out of a pocket into the loo, which is how most of my > company's DECT 'phones die). > > BTW, note that, for large campus sites, there's always something like > > Their CAL pricing for what is a basic SIP server is laughable, but the DECT > base stations and cellular controller are not too painful (i.e. it's a good > fit with fS ;). > > I have to add; SNOM DECT 'phones are OK-good, Panys you either like or > don't, but Polycom DECT phones are too toxic for landfill, IMHO. > > Problem I've seen with WIFi phones is battery life is not good, range is > less than DECT, and I haven't seen a *really* new design for quite a while. > Power is always going to be a problem, as WiFi Tx power is higher than > DECT, > and the link layer's more "chatty" on standby. > > all the best, > Lawrence > > On 5 May 2013, at 04:16, Mr Nathan Downes wrote: > > > I?ve used the Snom m3 and m9.. the Siemens c610ip and a510 are way > > better in feel/call quality and cheaper, not tried any others yet > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Luis Daniel Lucio Quiroz > > Sent: Sunday, 5 May 2013 12:44 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones > > > > > > > > I use at home snom m9 without problems > > > > El may 4, 2013 4:28 p.m., "Blake Priddy" > > escribi?: > > > > The SNOM DECT model is inexpensive and really good as well > > > > On May 4, 2013 3:23 PM, "Spencer Thomason" > > > > wrote: > > > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > > KX-TGP500s are great. > > > > > > > > -----Original message----- > > > > From: Sean Devoy > > To: FreeSWITCH-users at lists.freeswitch.org > > Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 > > Subject: [Freeswitch-users] WiFi IP VOIP SIP phones > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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-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://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-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://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-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://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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/0999399b/attachment-0001.html From mbodbg at gmx.net Sun May 5 22:58:33 2013 From: mbodbg at gmx.net (mbo) Date: Sun, 5 May 2013 20:58:33 +0200 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: Thanks for the answers. There is hardly any documentation of the PRIVATE_COMMAND event, can you give me some more details on it. What is it good for, when is it send, etc? If it is just an additional event including the channel data, where is the benefit using this event instead of set a custom variable first and then check this variable in the ChannelExecuteComplete event? Or am I missing something and it is possible to set a custom channel variable during SendMsg? Thanks Markus Am 03.05.2013 um 21:13 schrieb Anthony Minessale : > Compromise: > > Update to latest and sub to the "private_command" event, you should get back you own event with channel data merged in. > > > > On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: > > > On Fri, May 3, 2013 at 2:19 AM, mbo wrote: > I'm referring to a two years old bug report http://jira.freeswitch.org/browse/FS-1309. > > Is it in the meantime somehow possible to map reply to SendMsg in asyc mode? I'm wondering why this ticket has been closed as "Won't fix", in my opinion is an essential feature to handle events properly. > > If not, I want to implement the work around described in the ticket, to set a channel variable in a round trip before executing the real command. Do I need to wait for the ChannelExecuteComplete event for the Set command, or can I send my "real" command right away after the Set command? > To be safe you should verify that the set app actually ran before you implement the workaround. > -MC > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130505/b774ff19/attachment.html From Mailings at kh-dev.de Mon May 6 02:01:08 2013 From: Mailings at kh-dev.de (Klaus Hochlehnert) Date: Sun, 5 May 2013 22:01:08 +0000 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <128A8D2C-4B10-4503-A218-7D53C5E003BD@insensate.co.uk> <4d9901ce49ad$f1f8fe40$d5eafac0$@bizfocused.com> Message-ID: <6010458CBF7CF140ACE655A87A520739054387BC@srv01.khdev.corp> Hi Sean, I also did some tests with wireless phones. In my opinion you should really forget about the WiFi stuff and go for DECT. I never got a good voice quality and roaming between several (managed) access points was basically impossible when talking. My tests were based on Linksys and Gigaset WiFi phones. Not sure, but I think almost all WiFi SIP phones disappeared from the market (except some Android based smartphones which have a SIP client). About the DECT phones I found out that the Gigaset Pro series was performing better than the Snom DECT phones. I really love Snom phones, but their DECT products could never convince me. One of my customers use now a N510 IP PRO base and some SL610H PRO handsets. The spec says 6 phones and 4 lines. If you need more you should have a look at the N720 which should be a multi cell DECT system (I've never used it). But the website says 100 handsets, 20 base stations, overall 30 calls and 8 calls per base... Hope that helps... Klaus Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian Foster Gesendet: Sonntag, 5. Mai 2013 19:10 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] WiFi IP VOIP SIP phones I've dealt with a similar situation. I work for a property management company, and at one stage we were developing a strategy to basically resell voip service to tenants. Each would have a phone and prepay for minutes. We even tbought about having a kiosk to top up their account when needed by credit card. They could also top up by paying a little extra on top of their rent which would then be applied to their voip account. We ended up not doing it. And here's why: 1. WiFi/DECT phones are too expensive to replace. They would get beat up or lost. It's just too risky. On top of that, WiFi just isn't great at all for real-time media such as voip. You could do some tweaking to mitigate some of the issues, but it's still not worth it. Use DECT if you really want to go down this route. You could rent the phones, but due to our clientele at this property, it wasn't feasible. 2. We thought about using an FXS channel bank and running copper to every room. It takes care of the lost phone issue, and the tenant would be responsible for their phone because, well, they bought it. If the tenant wouldn't have been able to afford it, we could have set them up on a payment plan for a cordless phone. In total, we would have needed 16 channels, plus one for the office. Basically, a 24 port channel bank. They are relatively inexpensive, especially considering the use of wifi/dect phones. Unfortunately, there is not much in it if you want any sort of ROI. It would take a few years to get it, and by that time you wouldn't be far from replacing the equipment you originally installed. That's also being optimistic, as that estimate would assume 70% occupancy. Now there are several ways we could have pursued this further, but then the Obamaphone came out and by then, there was no demand. So, I'm not saying it will never work. It's just difficult and requires some major number crunching to make sure you stay above water. If there were 50 units, that would be a major help. But this property only has 16. If you need some help, I'd be happy to, ping me off list. I spent six months off and on studying the feasibility of such a venture. There's a lot of things you need to be aware of, and you need to know your client's habits. -BDF Sent from my Samsung Galaxy S4 On May 5, 2013 12:36 PM, "Sean Devoy" > wrote: Thanks to everyone who answered. I am dealing with a language barrier with this customer so bear with me. Now that I understand where these phones are actually going, I can refine my query. The basic concept is to change a boarding house with 12 rooms on a single analog line to VOIP with individual extensions. 6 of the rooms are wired or will be added easily. The other 6 rooms cannot be wire but will have WiFi access. It sounds like WiFi desk phones would be fine. Single Line/Extension is good. So, can 6 WiFi/DECT phones on distinct bases coexist in the same radio range? Now I see Blake has already replied and the SNOM can have a single base with up to 10 non-shared extensions! I guess that would be a strategically placed base and 6 charger/handsets. I shall ready up on SNOM. Would WiFi dongles on Desk phones work better for this venue? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lawrence Conroy Sent: Sunday, May 05, 2013 9:05 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones Hi Folks, +1 -- I use multiple Siemens S675IP DECT 'phones. Voice quality good (supports G722 and speakerphone is OK, unlike some), range is excellent, Battery life good, and they just work (assuming you don't drop them out of a pocket into the loo, which is how most of my company's DECT 'phones die). BTW, note that, for large campus sites, there's always something like Their CAL pricing for what is a basic SIP server is laughable, but the DECT base stations and cellular controller are not too painful (i.e. it's a good fit with fS ;). I have to add; SNOM DECT 'phones are OK-good, Panys you either like or don't, but Polycom DECT phones are too toxic for landfill, IMHO. Problem I've seen with WIFi phones is battery life is not good, range is less than DECT, and I haven't seen a *really* new design for quite a while. Power is always going to be a problem, as WiFi Tx power is higher than DECT, and the link layer's more "chatty" on standby. all the best, Lawrence On 5 May 2013, at 04:16, Mr Nathan Downes wrote: > I've used the Snom m3 and m9.. the Siemens c610ip and a510 are way > better in feel/call quality and cheaper, not tried any others yet > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Luis Daniel Lucio Quiroz > Sent: Sunday, 5 May 2013 12:44 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] WiFi IP VOIP SIP phones > > > > I use at home snom m9 without problems > > El may 4, 2013 4:28 p.m., "Blake Priddy" > > escribi?: > > The SNOM DECT model is inexpensive and really good as well > > On May 4, 2013 3:23 PM, "Spencer Thomason" > > > wrote: > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great. > > > > -----Original message----- > > From: Sean Devoy > > To: FreeSWITCH-users at lists.freeswitch.org > Sent: Sat, May 4, 2013 19:56:55 GMT+00:00 > Subject: [Freeswitch-users] WiFi IP VOIP SIP phones > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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-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://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-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://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-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://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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130505/0f7e01c9/attachment-0001.html From michel.brabants at gmail.com Mon May 6 02:40:42 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Mon, 6 May 2013 00:40:42 +0200 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> Message-ID: I believe the gigasetphones have their oen unique voipaccount Op 5 mei 2013 18:12 schreef "Sean Devoy" het volgende: > Spencer and Blake,**** > > I have seen DECT SIP phones and been fairly confused. Is that wired > Ethernet to the base and then a standard wireless home phone? My customer > wants these for rooms that have no wired Ethernet and need 6 distinct (not > shared) extensions. I don?t think you can have 6 DECT phones (to unique > bases) in one house can you?**** > > ** ** > > Thanks,**** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Blake Priddy > *Sent:* Saturday, May 04, 2013 4:25 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] WiFi IP VOIP SIP phones**** > > ** ** > > The SNOM DECT model is inexpensive and really good as well **** > > On May 4, 2013 3:23 PM, "Spencer Thomason" > wrote:**** > > Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic > KX-TGP500s are great.**** > > > > -----Original message-----**** > > *From: *Sean Devoy * > To: *FreeSWITCH-users at lists.freeswitch.org* > Sent: *Sat, May 4, 2013 19:56:55 GMT+00:00* > Subject: *[Freeswitch-users] WiFi IP VOIP SIP phones**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/31c12fe5/attachment.html From lconroy at insensate.co.uk Mon May 6 03:26:40 2013 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Mon, 6 May 2013 00:26:40 +0100 Subject: [Freeswitch-users] (not) WiFi IP VOIP SIP phones In-Reply-To: References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> Message-ID: Hi there, well, yes, there's gigaset.net, but it's not mandatory :). The 'phones/IP bases work fine with every sane SIP service I've thrown at them. (When you set the retail units up the first time, you can sign in to a dedicated gigaset.net account, but that's purely SIP-SIP only. Frankly, I've used gigaset.net account once, checked it worked, and disabled it again). 510 Pros support 4 concurrent calls; as mentioned, 720 pro/720 DM pro setups support LOTS of calls -- 4 is the minimum you get through a 720 base (if the phones are exchanging G722), but normal (G711/...) calls give you 8 per cell. The biggest problem I've had with a DECT setup is a couple of customers who needed (for security/legal reasons) to have a dead area with guaranteed *no* DECT coverage. It's amazing what range you get out of a DECT base when you don't want it. 25-50 metres indoors is normal, but it can easily be more :(. all the best, Lawrence On 5 May 2013, at 23:40, Michel Brabants wrote: > I believe the gigasetphones have their oen unique voipaccount > Op 5 mei 2013 18:12 schreef "Sean Devoy" het > volgende: > >> Spencer and Blake,**** >> >> I have seen DECT SIP phones and been fairly confused. Is that wired >> Ethernet to the base and then a standard wireless home phone? My customer >> wants these for rooms that have no wired Ethernet and need 6 distinct (not >> shared) extensions. I don?t think you can have 6 DECT phones (to unique >> bases) in one house can you?**** >> >> ** ** >> >> Thanks,**** >> >> Sean**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Blake Priddy >> *Sent:* Saturday, May 04, 2013 4:25 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] WiFi IP VOIP SIP phones**** >> >> ** ** >> >> The SNOM DECT model is inexpensive and really good as well **** >> >> On May 4, 2013 3:23 PM, "Spencer Thomason" >> wrote:**** >> >> Not strictly WiFi but if a DECT SIP phone will suffice, the Panasonic >> KX-TGP500s are great.**** >> >> >> >> -----Original message-----**** >> >> *From: *Sean Devoy * >> To: *FreeSWITCH-users at lists.freeswitch.org* >> Sent: *Sat, May 4, 2013 19:56:55 GMT+00:00* >> Subject: *[Freeswitch-users] WiFi IP VOIP SIP phones**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ashish at nms.co.in Mon May 6 10:19:55 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 6 May 2013 11:49:55 +0530 Subject: [Freeswitch-users] [ERR] ftdm_io.c:5740 Failed to load module type: libpri Message-ID: Hi, I am using FS on freeBSD with dahdi-libpri-ftdm stack. I am getting this error when I try to load mod_ftdm. dahdi_scan shows all channels ok. 2013-05-06 11:41:26.456705 [ERR] ftdm_io.c:5482 Error loading /usr/local/freeswitch/mod/ftmod_libpri.so [Cannot open "/usr/local/freeswitch/mod/ftmod_libpri.so"] 2013-05-06 11:41:26.456705 [ERR] ftdm_io.c:5740 Failed to load module type: libpri 2013-05-06 11:41:26.456705 [ERR] mod_freetdm.c:4076 Error configuring FreeTDM span FreeTDM1 Kindly help, Its really urgent. Regards, Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/d6042179/attachment.html From navnath.sonavne at yahoo.com Mon May 6 11:35:56 2013 From: navnath.sonavne at yahoo.com (Navnath Sonavne) Date: Mon, 6 May 2013 15:35:56 +0800 (SGT) Subject: [Freeswitch-users] ekiga to flex client call not happening Message-ID: <1367825756.24273.YahooMailNeo@web192204.mail.sg3.yahoo.com> Hi All, 1.Flex client to Ekiga :? I can make call from flex client given in freeswitch souce to ekiga(h323 protocol) softphone successfully. Flex client is registered on freeswitch as 1103.xml extension and ekiga phone is registered as 1104.xml extension. Here is dialplan for flex to ekiga call :? (These actions are written for testing purpose only not a permanent solution.) ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? 2.Ekiga to Flex client :? Now I am trying to make call from my Ekiga(h323 protocol) softphone to flex client given in freeswitch souce but it is failing every time. Flex client html page don't show any dialog box for incoming call but its? javascript code contains code for dialog for incoming call. Does that flex client is made only to make outging call?It can make outgoing call to ekiga phone. Isn't it made for incoming call? If that flex client is also made for incoming call too then please tell how to make call from ekiga to flex client. Here is dialplan for ekiga to flex call: (These actions are written for testing purpose only not a permanent solution.) ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? -Navnath. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/0076dffb/attachment-0001.html From smfarrukh at live.com Mon May 6 12:19:53 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Mon, 6 May 2013 08:19:53 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com>, , , , , Message-ID: I tried to configure rtp-timeout-sec in sofia.conf.xml as but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec -Steve On 3 May 2013 07:12, Farrukh Ali wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster at davri.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130506/7f948e1b/attachment.html From mehroz.ashraf85 at gmail.com Mon May 6 12:40:35 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 6 May 2013 01:40:35 -0700 (PDT) Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: <1367829635987-7590390.post@n2.nabble.com> Stucked at the same place! See here, http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-td6451091.html if there is anything common in your setup! I have been asked to take to logs of that , thats what i coudnt manage to do at yet! Looking forward for possible solution ! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590390.html Sent from the freeswitch-users mailing list archive at Nabble.com. From smfarrukh at live.com Mon May 6 13:44:17 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Mon, 6 May 2013 09:44:17 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1367829635987-7590390.post@n2.nabble.com> References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com>, , , , , , , <1367829635987-7590390.post@n2.nabble.com> Message-ID: yes, same problem is with me, it looks like we are missing something, > Date: Mon, 6 May 2013 01:40:35 -0700 > From: mehroz.ashraf85 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Stucked at the same place! > See here, > http://freeswitch-users.2379917.n2.nabble.com/Check-if-UA-is-still-there-td6451091.html > > if there is anything common in your setup! I have been asked to take to logs > of that , thats what i coudnt manage to do at yet! Looking forward for > possible solution ! > > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590390.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://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/20130506/9fb5ec93/attachment.html From steveayre at gmail.com Mon May 6 14:08:17 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 6 May 2013 11:08:17 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: Did you ensure that the parameter was set on the sofia profile? By default they're not contained in the sofia.conf.xml file but in the files it pulls in via the X-PRE-PROCESS include. It will not work if you set it in the section. Also did you try restarting FS after setting the parameter (restarting the profile should also work, but restarting the process if you can is a simple way to be 100% sure it's reloaded the entire configuration). -Steve On 6 May 2013 09:19, Farrukh Ali wrote: > I tried to configure rtp-timeout-sec in sofia.conf.xml as > > > > but did not get any change, dead channels don't get hung up! > Please reply > > Regards, > Muhammad Farrukh > > ------------------------------ > Date: Fri, 3 May 2013 09:39:14 -0500 > From: anthony.minessale at gmail.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Asterisk uses the same 2 methods. The difference is probably that one or > both are set by default or are implicit from some other setting. > On May 3, 2013 3:01 AM, "Steven Ayre" wrote: > > If you're on VoIP and an endpoint completely disappears (eg wifi down) > then the is absolutely no way for the other endpoint to detect that. That > goes whatever software you're using. > > It's impossible for the down endpoint to tell the other that it's down, > since it's down. Lack of messages also isn't enough to know it's down - > it's normal during most of the SIP call for there to be no SIP packets, and > RTP is designed to cope with packet loss. > > Absolutely the only way to detect it is to spot when you haven't heard > from them for a while and timeout. Using SIP/UDP that's either in > signalling with session timers (send a SIP packet, wait for a reply, > timeout if no response within N seconds), or in media with RTP timers > (hangup call if no RTP received for N seconds). > > To use session timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout > > Or to use RTP timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec > > -Steve > > > > > On 3 May 2013 07:12, Farrukh Ali wrote: > > Yes BDF, you are right, but I have been using Asterisk for a while and > never faced such problem, there has to be some mechanism Freeswitch is > lagging, beside this Freeswitch is quiet flexible to use due to its unique > XML configuration style, > And thanks for response > > Regards > Muhammad Farrukh > > ------------------------------ > Date: Thu, 2 May 2013 08:51:42 -0400 > From: bdfoster at davri.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Just a note: > > In your setup, a timeout if some sort is your only option. It may not be > efficient, it may not be perfect, but neither are your WiFi/Mobile clients > apparently. > > -BDF > On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: > > Thanks everyone, and Steve timers might not be efficient solution but for > now it will be good for testing purpose, could you please tell me exactly > which file and parameter should I look for to change, is this configuration > located in sofia.conf.xml ? kindly reply! > and thanks to all again! > > Regards, > Muhammad Farrukh > > ------------------------------ > From: steveayre at gmail.com > Date: Tue, 30 Apr 2013 17:20:36 +0100 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > SIP over UDP has no way of knowing the other side has lost connectivity > (eg wifi down). Sip droid will know the local network is down and be able > to hang up the call because of that. FreeSWITCH has no such knowledge, so > won't know the call is hung up and will continue sending RTP. > > 3 possibilities are: > 1) Use session timers > 2) There is a parameter to hang up a call is no RTP has been received for > a certain period of time > 3) Use SIP/TCP, keepalives might help (but the default is measured in > hours) > > Note that all of these are timeouts - you simply won't be able to know the > call has ended immediately. > > Steve > > > > On 30 Apr 2013, at 08:33, Farrukh Ali wrote: > > Hi, > > I need some help, there is an issue in Freeswitch with SIP clients, when a > call is not hung up properly Freeswitch does not close its RTP channels, > and it starts to get load on bandwidth after too many dead channels are > generated, Improper hang up means if during call sip client went out of the > range of WiFi, the client i.e SipDroid shows call hang up but freeswitch > continues to send voice packets. Kindly reply > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/5a38765f/attachment-0001.html From smfarrukh at live.com Mon May 6 15:41:35 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Mon, 6 May 2013 11:41:35 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com>, , , , , , , Message-ID: Hi Steve, So, should I put it in profile tag? From: steveayre at gmail.com Date: Mon, 6 May 2013 11:08:17 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Did you ensure that the parameter was set on the sofia profile? By default they're not contained in the sofia.conf.xml file but in the files it pulls in via the X-PRE-PROCESS include. It will not work if you set it in the section. Also did you try restarting FS after setting the parameter (restarting the profile should also work, but restarting the process if you can is a simple way to be 100% sure it's reloaded the entire configuration). -Steve On 6 May 2013 09:19, Farrukh Ali wrote: I tried to configure rtp-timeout-sec in sofia.conf.xml as but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec -Steve On 3 May 2013 07:12, Farrukh Ali wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster at davri.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130506/8f0c78ad/attachment-0001.html From thomas.peterseil at gmail.com Mon May 6 16:06:48 2013 From: thomas.peterseil at gmail.com (thomas peterseil) Date: Mon, 6 May 2013 14:06:48 +0200 Subject: [Freeswitch-users] FS transcoding Message-ID: hello, i am running a new FS and i have two sip clients registered, one which supports G.722 and the other one supports G.711. i am still running mostly the default config, both codecs are listed when i do a show codecs. when i call now from 1000 to 1001 with different codecs it doesnt work. but i can call from both clients with different codecs to a 3300 conference. is there something i have to enable in the default config to activate transcoding? thanks for your help! thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/7ad24dd1/attachment.html From bpriddy at bryantschools.org Mon May 6 16:29:03 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Mon, 6 May 2013 07:29:03 -0500 Subject: [Freeswitch-users] FS transcoding In-Reply-To: References: Message-ID: Maybe you need to list those in your vars.xml On May 6, 2013 7:12 AM, "thomas peterseil" wrote: > hello, > i am running a new FS and i have two sip clients registered, one which > supports G.722 and the other one supports G.711. i am still running mostly > the default config, both codecs are listed when i do a show codecs. when i > call now from 1000 to 1001 with different codecs it doesnt work. but i can > call from both clients with different codecs to a 3300 conference. > is there something i have to enable in the default config to activate > transcoding? > > thanks for your help! > > thomas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/e066d81c/attachment.html From thomas.peterseil at gmail.com Mon May 6 16:51:38 2013 From: thomas.peterseil at gmail.com (thomas peterseil) Date: Mon, 6 May 2013 14:51:38 +0200 Subject: [Freeswitch-users] FS transcoding In-Reply-To: References: Message-ID: thank you very much for your help, but they are enabled in the vars.xml any other suggestions? thanks, thomas 2013/5/6 Blake Priddy > Maybe you need to list those in your vars.xml > On May 6, 2013 7:12 AM, "thomas peterseil" > wrote: > >> hello, >> i am running a new FS and i have two sip clients registered, one which >> supports G.722 and the other one supports G.711. i am still running mostly >> the default config, both codecs are listed when i do a show codecs. when i >> call now from 1000 to 1001 with different codecs it doesnt work. but i can >> call from both clients with different codecs to a 3300 conference. >> is there something i have to enable in the default config to activate >> transcoding? >> >> thanks for your help! >> >> thomas >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/e483c216/attachment.html From steveayre at gmail.com Mon May 6 17:45:31 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 6 May 2013 14:45:31 +0100 Subject: [Freeswitch-users] FS transcoding In-Reply-To: References: Message-ID: Can you post a debug-level log of a call between the extensions? -Steve On 6 May 2013 13:51, thomas peterseil wrote: > thank you very much for your help, but they are enabled in the vars.xml > > > > > any other suggestions? > > thanks, > > thomas > > > 2013/5/6 Blake Priddy > >> Maybe you need to list those in your vars.xml >> On May 6, 2013 7:12 AM, "thomas peterseil" >> wrote: >> >>> hello, >>> i am running a new FS and i have two sip clients registered, one which >>> supports G.722 and the other one supports G.711. i am still running mostly >>> the default config, both codecs are listed when i do a show codecs. when i >>> call now from 1000 to 1001 with different codecs it doesnt work. but i can >>> call from both clients with different codecs to a 3300 conference. >>> is there something i have to enable in the default config to activate >>> transcoding? >>> >>> thanks for your help! >>> >>> thomas >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/ac4b9f9a/attachment-0001.html From steveayre at gmail.com Mon May 6 17:47:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 6 May 2013 14:47:04 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com> Message-ID: Note the settings tag within profile. The same goes for the other parameters previously mentioned. -Steve On 6 May 2013 12:41, Farrukh Ali wrote: > Hi Steve, > > So, should I put it in profile tag? > > ------------------------------ > From: steveayre at gmail.com > Date: Mon, 6 May 2013 11:08:17 +0100 > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Did you ensure that the parameter was set on the sofia profile? By default > they're not contained in the sofia.conf.xml file but in the files it pulls > in via the X-PRE-PROCESS include. > > It will not work if you set it in the section. > > Also did you try restarting FS after setting the parameter (restarting the > profile should also work, but restarting the process if you can is a simple > way to be 100% sure it's reloaded the entire configuration). > > -Steve > > > > On 6 May 2013 09:19, Farrukh Ali wrote: > > I tried to configure rtp-timeout-sec in sofia.conf.xml as > > > > but did not get any change, dead channels don't get hung up! > Please reply > > Regards, > Muhammad Farrukh > > ------------------------------ > Date: Fri, 3 May 2013 09:39:14 -0500 > From: anthony.minessale at gmail.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Asterisk uses the same 2 methods. The difference is probably that one or > both are set by default or are implicit from some other setting. > On May 3, 2013 3:01 AM, "Steven Ayre" wrote: > > If you're on VoIP and an endpoint completely disappears (eg wifi down) > then the is absolutely no way for the other endpoint to detect that. That > goes whatever software you're using. > > It's impossible for the down endpoint to tell the other that it's down, > since it's down. Lack of messages also isn't enough to know it's down - > it's normal during most of the SIP call for there to be no SIP packets, and > RTP is designed to cope with packet loss. > > Absolutely the only way to detect it is to spot when you haven't heard > from them for a while and timeout. Using SIP/UDP that's either in > signalling with session timers (send a SIP packet, wait for a reply, > timeout if no response within N seconds), or in media with RTP timers > (hangup call if no RTP received for N seconds). > > To use session timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout > > Or to use RTP timers: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec > > -Steve > > > > > On 3 May 2013 07:12, Farrukh Ali wrote: > > Yes BDF, you are right, but I have been using Asterisk for a while and > never faced such problem, there has to be some mechanism Freeswitch is > lagging, beside this Freeswitch is quiet flexible to use due to its unique > XML configuration style, > And thanks for response > > Regards > Muhammad Farrukh > > ------------------------------ > Date: Thu, 2 May 2013 08:51:42 -0400 > From: bdfoster at davri.com > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Just a note: > > In your setup, a timeout if some sort is your only option. It may not be > efficient, it may not be perfect, but neither are your WiFi/Mobile clients > apparently. > > -BDF > On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: > > Thanks everyone, and Steve timers might not be efficient solution but for > now it will be good for testing purpose, could you please tell me exactly > which file and parameter should I look for to change, is this configuration > located in sofia.conf.xml ? kindly reply! > and thanks to all again! > > Regards, > Muhammad Farrukh > > ------------------------------ > From: steveayre at gmail.com > Date: Tue, 30 Apr 2013 17:20:36 +0100 > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > SIP over UDP has no way of knowing the other side has lost connectivity > (eg wifi down). Sip droid will know the local network is down and be able > to hang up the call because of that. FreeSWITCH has no such knowledge, so > won't know the call is hung up and will continue sending RTP. > > 3 possibilities are: > 1) Use session timers > 2) There is a parameter to hang up a call is no RTP has been received for > a certain period of time > 3) Use SIP/TCP, keepalives might help (but the default is measured in > hours) > > Note that all of these are timeouts - you simply won't be able to know the > call has ended immediately. > > Steve > > > > On 30 Apr 2013, at 08:33, Farrukh Ali wrote: > > Hi, > > I need some help, there is an issue in Freeswitch with SIP clients, when a > call is not hung up properly Freeswitch does not close its RTP channels, > and it starts to get load on bandwidth after too many dead channels are > generated, Improper hang up means if during call sip client went out of the > range of WiFi, the client i.e SipDroid shows call hang up but freeswitch > continues to send voice packets. Kindly reply > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/3c0e1e69/attachment-0001.html From mehroz.ashraf85 at gmail.com Mon May 6 17:54:08 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 6 May 2013 06:54:08 -0700 (PDT) Subject: [Freeswitch-users] Any one with bluebox working? Message-ID: <1367848448292-7590398.post@n2.nabble.com> Hi, I have recently installed bluebox for FS GUI ! I have no issue with audio calling, but video calling is messed up. Codecs have been changed and the same set of configuration works on Local LAN (no NAT), however, it is troublesome with NAT! Tried all combination of NAT settngs in SIP profile setting but none of it works. Please respond if anyone undergone with this issue. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Any-one-with-bluebox-working-tp7590398.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at davri.com Mon May 6 19:35:57 2013 From: bdfoster at davri.com (Brian Foster) Date: Mon, 6 May 2013 11:35:57 -0400 Subject: [Freeswitch-users] ekiga to flex client call not happening In-Reply-To: <1367825756.24273.YahooMailNeo@web192204.mail.sg3.yahoo.com> References: <1367825756.24273.YahooMailNeo@web192204.mail.sg3.yahoo.com> Message-ID: Need a console showing where the call fails. Please use pastebin: http://pastebin.freeswitch.org On May 6, 2013 3:42 AM, "Navnath Sonavne" wrote: > Hi All, > > 1.Flex client to Ekiga : > I can make call from flex client given in freeswitch souce to ekiga(h323 > protocol) softphone successfully. > Flex client is registered on freeswitch as 1103.xml extension and ekiga > phone is registered as 1104.xml extension. > > Here is dialplan for flex to ekiga call : > (These actions are written for testing purpose only not a permanent > solution.) > > > > data="effective_caller_id_number=${dialed_extension}"/> > > > > > > > > > > > 2.Ekiga to Flex client : > Now I am trying to make call from my Ekiga(h323 protocol) softphone to > flex client given in freeswitch souce but > it is failing every time. > > Flex client html page don't show any dialog box for incoming call but its > javascript code contains code for dialog for incoming call. > Does that flex client is made only to make outging call?It can make > outgoing call to ekiga phone. > Isn't it made for incoming call? > If that flex client is also made for incoming call too then please tell > how to make call from ekiga to flex client. > > > Here is dialplan for ekiga to flex call: > (These actions are written for testing purpose only not a permanent > solution.) > > > > data="effective_caller_id_number=${dialed_extension}"/> > > > > > data="${rtmp_contact($${rtmp_profile}/${dialed_extension}@ > $${domain_name})}"/> > > > > > > -Navnath. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/43dc6e89/attachment.html From msc at freeswitch.org Mon May 6 19:36:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 May 2013 08:36:18 -0700 Subject: [Freeswitch-users] FS transcoding In-Reply-To: References: Message-ID: On Mon, May 6, 2013 at 5:51 AM, thomas peterseil wrote: > thank you very much for your help, but they are enabled in the vars.xml > > > > Do you want OPUS, SILK, and G7221 to be preferred over G722 and PCMU? Just curious because you have all of those listed before the two codecs you actually meant to use. In any case, please get the debug log for the call between the phones and put it on pastebin.freeswitch.org so that we can see what's happening. -MC > any other suggestions? > > thanks, > > thomas > > > 2013/5/6 Blake Priddy > >> Maybe you need to list those in your vars.xml >> On May 6, 2013 7:12 AM, "thomas peterseil" >> wrote: >> >>> hello, >>> i am running a new FS and i have two sip clients registered, one which >>> supports G.722 and the other one supports G.711. i am still running mostly >>> the default config, both codecs are listed when i do a show codecs. when i >>> call now from 1000 to 1001 with different codecs it doesnt work. but i can >>> call from both clients with different codecs to a 3300 conference. >>> is there something i have to enable in the default config to activate >>> transcoding? >>> >>> thanks for your help! >>> >>> thomas >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/14b41c6d/attachment-0001.html From dwilkie at gmail.com Mon May 6 21:10:24 2013 From: dwilkie at gmail.com (David Wilkie) Date: Tue, 7 May 2013 00:10:24 +0700 Subject: [Freeswitch-users] Early Media indication from B in CHANNEL_STATE event Message-ID: Hi, Is there a way to get an early media indication from a B-leg in the CHANNEL_STATE event. According to http://wiki.freeswitch.org/wiki/Event_list#CHANNEL_STATE there doesn't seem to be a variable which indicates early media from B. We need to use this information to figure out whether to join B to A or generate early media and join on answer. If there is no way to get this info can we query it using Event Socket? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/ef95ddeb/attachment.html From brian at freeswitch.org Mon May 6 21:17:01 2013 From: brian at freeswitch.org (Brian West) Date: Mon, 6 May 2013 12:17:01 -0500 Subject: [Freeswitch-users] WiFi IP VOIP SIP phones In-Reply-To: <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> References: <495e01ce4900$d14571e0$73d055a0$@bizfocused.com> <16ef6ab1-4ac2-42c6-b466-80e9515f1f7d@blur> <4d7001ce49ab$0f5b9ae0$2e12d0a0$@bizfocused.com> Message-ID: I personally have a Polycom KWS300 with 4 handsets. I love them. They feel kinda light but are well made, I use it in my home and office. They have repeaters for the kws300 too. I may look into getting a KWS6000 to play with. On May 5, 2013, at 11:10 AM, Sean Devoy wrote: > Spencer and Blake, > I have seen DECT SIP phones and been fairly confused. Is that wired Ethernet to the base and then a standard wireless home phone? My customer wants these for rooms that have no wired Ethernet and need 6 distinct (not shared) extensions. I don?t think you can have 6 DECT phones (to unique bases) in one house can you? > > Thanks, > Sean From lloyd.aloysius at gmail.com Mon May 6 21:50:48 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Mon, 6 May 2013 13:50:48 -0400 Subject: [Freeswitch-users] pickup endpoint - Break the Simultaneous Ring Message-ID: Hello All The following break the Simultaneous Ring.User registered two different devices. Only one Device ring after add the *,pickup/dave at alcan.mydomain.ca * *bridge_data: * {sip_invite_domain=alcan.mydomain.ca ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca }[leg_timeout=20]user/dave at alcan.mydomain.ca*,pickup/dave at alcan.mydomain.ca* Thanks LLoyd * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/c72209b3/attachment.html From sertys at gmail.com Mon May 6 23:12:21 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Mon, 6 May 2013 21:12:21 +0200 Subject: [Freeswitch-users] ZRTP issue being hard to track In-Reply-To: References: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> Message-ID: This param did it. Now it runs in zrtp proxy mode. Thanks a ton for the tip. Someone needs to put the param in the wiki. On May 5, 2013 2:49 PM, "Daniel Ivanov" wrote: > I have missed this param, cannot wait to try on monday. Will report if > that's it. > On May 4, 2013 12:19 AM, "royj" wrote: > >> May be in profile >> would change picture >> >> On Fri, 3 May 2013 10:33:09 +0200 >> Daniel Ivanov wrote: >> >> > I make currented the production fs a few weeks ago, because i've been >> > unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup >> ). >> > Trusted mitm works, but once i enable proxy-media the UAs ignore the >> > zrtp-hash. One of them always says 'other side doesnt seem to support >> > zrto'. I talked on irc to steevenielson to ask him any hints and he >> pointed >> > the only difference between me and ostel.me setup is the rather old >> version >> > he has. Could you elaborate with me on this? >> >> >> -- >> royj >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130506/c4645acc/attachment.html From jmesquita at freeswitch.org Mon May 6 23:25:39 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 6 May 2013 16:25:39 -0300 Subject: [Freeswitch-users] ZRTP issue being hard to track In-Reply-To: References: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> Message-ID: You could be the lucky someone, how about that?! :D Jo?o Mesquita FreeSWITCH? Solutions On Mon, May 6, 2013 at 4:12 PM, Daniel Ivanov wrote: > This param did it. Now it runs in zrtp proxy mode. Thanks a ton for the > tip. Someone needs to put the param in the wiki. > On May 5, 2013 2:49 PM, "Daniel Ivanov" wrote: > >> I have missed this param, cannot wait to try on monday. Will report if >> that's it. >> On May 4, 2013 12:19 AM, "royj" wrote: >> >>> May be in profile >>> would change picture >>> >>> On Fri, 3 May 2013 10:33:09 +0200 >>> Daniel Ivanov wrote: >>> >>> > I make currented the production fs a few weeks ago, because i've been >>> > unable to run a zrtp call between 2 csipsimple UAs( it is an ostn >>> setup ). >>> > Trusted mitm works, but once i enable proxy-media the UAs ignore the >>> > zrtp-hash. One of them always says 'other side doesnt seem to support >>> > zrto'. I talked on irc to steevenielson to ask him any hints and he >>> pointed >>> > the only difference between me and ostel.me setup is the rather old >>> version >>> > he has. Could you elaborate with me on this? >>> >>> >>> -- >>> royj >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130506/1633222c/attachment.html From marketing at cluecon.com Tue May 7 00:50:17 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 6 May 2013 13:50:17 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Happy first May Monday to all. We are please to let everyone know that the new FreeSWITCH bookis nearing completion. Each day the Packt editors have been sending us revised and re-revised chapters to polish off. If we keep up our current pace then it is quite likely the book will be available in early June. We hope you enjoy it. On last week's call we enjoyed a nice presentation from Omar over at OrecX who showed us some of the useful features of their call recording solutions. They have an open source version as well as commercial versions. If you are looking for a call recording solution then we recommend that you review what OrecX has to offer. This week we continue in our series of presentations on FreeSWITCH-compatible software applications. We look forward to having Dan Bogos from CGRateSjoin us to talk about their project. Click here to get a quick look at what CGRateS does, and then make plans to join our call on Wednesday. Regarding ClueCon 2013 we would like to remind everyone that we are still accepting talk proposals. If you have a talk idea please let us know. In the meantime feel free to get registered and book your hotel room at the Hyatt. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130506/7a7163bc/attachment-0001.html From lists at kavun.ch Tue May 7 06:10:20 2013 From: lists at kavun.ch (Emrah) Date: Mon, 6 May 2013 19:10:20 -0700 Subject: [Freeswitch-users] Bind to multiple ports with Sofia In-Reply-To: <3096D814-BFB0-4D7E-8320-BC676C60C21B@freeswitch.org> References: <6678CCF7-CD89-4B9B-8A19-3951F8975BB6@kavun.ch> <3096D814-BFB0-4D7E-8320-BC676C60C21B@freeswitch.org> Message-ID: You guys are great it worked! Thanks a million On Apr 30, 2013, at 6:31 PM, Ken Rice wrote: > theres a way to do it, i think it might be */user at domain i forget tho.... > > Ken > Sent from my iPad > > On Apr 30, 2013, at 19:04, Emrah wrote: > >> Hi Ken, >> >> Even with the force-domain option, I can register on a different port but will not be found via sofia_contact user at domain. >> My impression is that only one profile can alter the register table. If the 5070 profile is up first, than my registrations will work on port 5070 only. If it's the main one on 5060 that comes up first, my contacts on 5070 disappear. >> >> Any advice would be very much appreciated. >> >> Thanks a bunch for your help, >> Emrah >> On Apr 28, 2013, at 7:14 AM, Ken Rice wrote: >> >>> You can have any number of domains on different profiles... >>> >>> If you just need 1 domain everywhere check the force domain options on the >>> sip profile configs (I believe they are used in the default example configs >>> >>> >>> On 4/28/13 8:42 AM, "Emrah" wrote: >>> >>>> Hi guys, >>>> >>>> I need to bind to multiple ports with Sofia. If I start one profile per port, >>>> I cannot get a consistent registration table working. A domain can only be >>>> aliased to one profile if my understanding is correct. >>>> >>>> If you have a working set up, I'd appreciate some advice. >>>> >>>> Thanks, >>>> Emrah >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> -- >>> Ken >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> irc.freenode.net #freeswitch >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From smfarrukh at live.com Tue May 7 09:34:57 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Tue, 7 May 2013 05:34:57 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: , <7CFD9622-E996-4099-85D2-025D13FED098@gmail.com>, , , , , , , , , Message-ID: Hey Steven, Thank you very much It works, Freeswitch stills shows an ongoing call, but stops its RTP traffic which was my concern, Thanks again And Mehroz you should try this as well! Regards Muhammad Farrukh From: steveayre at gmail.com Date: Mon, 6 May 2013 14:47:04 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Note the settings tag within profile. The same goes for the other parameters previously mentioned. -Steve On 6 May 2013 12:41, Farrukh Ali wrote: Hi Steve, So, should I put it in profile tag? From: steveayre at gmail.com Date: Mon, 6 May 2013 11:08:17 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Did you ensure that the parameter was set on the sofia profile? By default they're not contained in the sofia.conf.xml file but in the files it pulls in via the X-PRE-PROCESS include. It will not work if you set it in the section. Also did you try restarting FS after setting the parameter (restarting the profile should also work, but restarting the process if you can is a simple way to be 100% sure it's reloaded the entire configuration). -Steve On 6 May 2013 09:19, Farrukh Ali wrote: I tried to configure rtp-timeout-sec in sofia.conf.xml as but did not get any change, dead channels don't get hung up! Please reply Regards, Muhammad Farrukh Date: Fri, 3 May 2013 09:39:14 -0500 From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Asterisk uses the same 2 methods. The difference is probably that one or both are set by default or are implicit from some other setting. On May 3, 2013 3:01 AM, "Steven Ayre" wrote: If you're on VoIP and an endpoint completely disappears (eg wifi down) then the is absolutely no way for the other endpoint to detect that. That goes whatever software you're using. It's impossible for the down endpoint to tell the other that it's down, since it's down. Lack of messages also isn't enough to know it's down - it's normal during most of the SIP call for there to be no SIP packets, and RTP is designed to cope with packet loss. Absolutely the only way to detect it is to spot when you haven't heard from them for a while and timeout. Using SIP/UDP that's either in signalling with session timers (send a SIP packet, wait for a reply, timeout if no response within N seconds), or in media with RTP timers (hangup call if no RTP received for N seconds). To use session timers:http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout Or to use RTP timers: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec -Steve On 3 May 2013 07:12, Farrukh Ali wrote: Yes BDF, you are right, but I have been using Asterisk for a while and never faced such problem, there has to be some mechanism Freeswitch is lagging, beside this Freeswitch is quiet flexible to use due to its unique XML configuration style, And thanks for response Regards Muhammad Farrukh Date: Thu, 2 May 2013 08:51:42 -0400 From: bdfoster at davri.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch Just a note: In your setup, a timeout if some sort is your only option. It may not be efficient, it may not be perfect, but neither are your WiFi/Mobile clients apparently. -BDF On May 2, 2013 7:09 AM, "Farrukh Ali" wrote: Thanks everyone, and Steve timers might not be efficient solution but for now it will be good for testing purpose, could you please tell me exactly which file and parameter should I look for to change, is this configuration located in sofia.conf.xml ? kindly reply! and thanks to all again! Regards, Muhammad Farrukh From: steveayre at gmail.com Date: Tue, 30 Apr 2013 17:20:36 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Dead channels in freeswitch SIP over UDP has no way of knowing the other side has lost connectivity (eg wifi down). Sip droid will know the local network is down and be able to hang up the call because of that. FreeSWITCH has no such knowledge, so won't know the call is hung up and will continue sending RTP. 3 possibilities are:1) Use session timers2) There is a parameter to hang up a call is no RTP has been received for a certain period of time3) Use SIP/TCP, keepalives might help (but the default is measured in hours) Note that all of these are timeouts - you simply won't be able to know the call has ended immediately. Steve On 30 Apr 2013, at 08:33, Farrukh Ali wrote: Hi, I need some help, there is an issue in Freeswitch with SIP clients, when a call is not hung up properly Freeswitch does not close its RTP channels, and it starts to get load on bandwidth after too many dead channels are generated, Improper hang up means if during call sip client went out of the range of WiFi, the client i.e SipDroid shows call hang up but freeswitch continues to send voice packets. Kindly reply _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130507/347e54e1/attachment-0001.html From mehroz.ashraf85 at gmail.com Tue May 7 10:15:18 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Mon, 6 May 2013 23:15:18 -0700 (PDT) Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: Message-ID: <1367907318527-7590411.post@n2.nabble.com> I have already got that at the right place Farrukh! But unable to get the desired result! calls does not hangs up , which i monitored for 5 min and have to do "hupall" to get it fixed. Is this parameter have any dependency over any other parameter , liker timers? as i have disabled session timer param name="enable-timer" value="false", but AFAIK session timer and rtp-timers are two different entities! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.html Sent from the freeswitch-users mailing list archive at Nabble.com. From smfarrukh at live.com Tue May 7 13:12:07 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Tue, 7 May 2013 09:12:07 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1367907318527-7590411.post@n2.nabble.com> References: , , , , , , , , , , <1367907318527-7590411.post@n2.nabble.com> Message-ID: Hey Mehroz, The problem with this timer is it only STOPS the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires! Regards Muhammad Farrukh > Date: Mon, 6 May 2013 23:15:18 -0700 > From: mehroz.ashraf85 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > I have already got that at the right place Farrukh! > > > > > > > > But unable to get the desired result! calls does not hangs up , which i > monitored for 5 min and have to do "hupall" to get it fixed. > > Is this parameter have any dependency over any other parameter , liker > timers? as i have disabled session timer > param name="enable-timer" value="false", but AFAIK session timer and > rtp-timers are two different entities! > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.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://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/20130507/3e4b8eb1/attachment.html From steveayre at gmail.com Tue May 7 13:16:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 7 May 2013 10:16:21 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1367907318527-7590411.post@n2.nabble.com> References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: > But unable to get the desired result! calls does not hangs up , which i > monitored for 5 min and have to do "hupall" to get it fixed. Capture a trace of the call using Wireshark. Are you sure there's no RTP packets during that period? Also remember that the config change needs to be picked up by restarting the FreeSWITCH profile (the easiest way to be 100% sure is just to restart FS). Sharing your configuration could also be useful, in case there's a typo. > Is this parameter have any dependency over any other parameter , liker > timers? as i have disabled session timer > param name="enable-timer" value="false", but AFAIK session timer and > rtp-timers are two different entities! Correct, they're different. Session timers are a request/response in the SIP signalling layer. RTP timers are simply no RTP media packets received for a given period of time. There's no dependancy between them. You can use either none/one/the other/both. Incidentally the reason to prefer SIP session timers to RTP timers is that not all calls will send RTP. For example FS won't by default when it's only recording something (voicemail): http://wiki.freeswitch.org/wiki/Variable_record_waste_resources -Steve From ashish at nms.co.in Tue May 7 13:54:19 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 7 May 2013 15:24:19 +0530 Subject: [Freeswitch-users] urgent!!!! FS crashes on inbound calls Message-ID: Hi, I am running freeswitch on FreeBSD system with dahdi-libpri-freetdm stack. Here a serious issue is coming. Whenever I make an inbound call, FreeSWITCH just crashes with no errors in logs. I am attaching the part of freeswitch.log for reference. Please throw some light. Regards, Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/1ba28b75/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch_new.log Type: application/octet-stream Size: 21440 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/1ba28b75/attachment-0001.obj From shaheryarkh at gmail.com Tue May 7 14:00:13 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Tue, 7 May 2013 12:00:13 +0200 Subject: [Freeswitch-users] urgent!!!! FS crashes on inbound calls In-Reply-To: References: Message-ID: If it generates dump core then please share backtrace otherwise enable dump core and provide backtrace. Thank you. On Tue, May 7, 2013 at 11:54 AM, Ashish gautam wrote: > Hi, > > I am running freeswitch on FreeBSD system with dahdi-libpri-freetdm stack. > Here a serious issue is coming. Whenever I make an inbound call, FreeSWITCH > just crashes with no errors in logs. I am attaching the part of > freeswitch.log for reference. > > Please throw some light. > > Regards, > Ashish > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/d625b135/attachment.html From GB at cm.nl Tue May 7 16:04:11 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 7 May 2013 14:04:11 +0200 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables Message-ID: Hello, When using the Originate command, is it possible to generate a dynamic sequence of actions to take place when the user answers the call? For example: Originate command is called with some arguments in this specific order; playback, transfer, playback. FreeSwitch should first play something, then transfer to some destination, and after returning from the transfer, play another message. The sequence of these actions is defined by the external application which is calling the Originate command. If this is possible, what is the best way to accomplish this? I have no experience with FreeSwitch, just doing some research. Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/cb6dc367/attachment.html From steveayre at gmail.com Tue May 7 16:20:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 7 May 2013 13:20:18 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: No, it should actually hang up the call... Steve On 7 May 2013, at 10:12, Farrukh Ali wrote: > Hey Mehroz, > > The problem with this timer is it only STOPS the RTP traffic going from FS to SIPClient by detecting RTP is not received from the SIPClient for certain time(you set), It doesn't kill the session as you can check it by pressing F4, you need to hupall all the time, In Wireshark you could see that RTP stream is stopped after the timer expires! > > Regards > Muhammad Farrukh > > > > > Date: Mon, 6 May 2013 23:15:18 -0700 > > From: mehroz.ashraf85 at gmail.com > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > > > I have already got that at the right place Farrukh! > > > > > > > > > > > > > > > > But unable to get the desired result! calls does not hangs up , which i > > monitored for 5 min and have to do "hupall" to get it fixed. > > > > Is this parameter have any dependency over any other parameter , liker > > timers? as i have disabled session timer > > param name="enable-timer" value="false", but AFAIK session timer and > > rtp-timers are two different entities! > > > > > > > > -- > > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.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://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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/ebcb1d3e/attachment.html From clan at wheel.dk Tue May 7 13:57:03 2013 From: clan at wheel.dk (Claus Andersen) Date: Tue, 7 May 2013 11:57:03 +0200 (CEST) Subject: [Freeswitch-users] ZRTP issue being hard to track In-Reply-To: References: <20130504011559.941910bf5283ce6af548bc5b@yandex.ru> Message-ID: On Mon, 6 May 2013, Jo?o Mesquita wrote: > You could be the lucky someone, how about that?! :D I felt lucky - so... https://wiki.freeswitch.org/wiki/ZRTP#ZRTP_passthru https://wiki.freeswitch.org/wiki/Proxy_Media#Proxy_media_does_not_work_with_ZRTP_trusted_MITM Please feel free to make corrections. I rarely know what I am doing... Kind Regards, Claus Andersen > On Mon, May 6, 2013 at 4:12 PM, Daniel Ivanov wrote: > > This param did it. Now it runs in zrtp proxy mode. Thanks a ton for the tip. Someone needs to put > the param in the wiki. > > On May 5, 2013 2:49 PM, "Daniel Ivanov" wrote: > > I have missed this param, cannot wait to try on monday. Will report if that's it. > > On May 4, 2013 12:19 AM, "royj" wrote: > May be in profile would > change picture > > On Fri, 3 May 2013 10:33:09 +0200 > Daniel Ivanov wrote: > > > I make currented the production fs a few weeks ago, because i've been > > unable to run a zrtp call between 2 csipsimple UAs( it is an ostn setup ). > > Trusted mitm works, but once i enable proxy-media the UAs ignore the > > zrtp-hash. One of them always says 'other side doesnt seem to support > > zrto'. I talked on irc to steevenielson to ask him any hints and he > pointed > > the only difference between me and ostel.me setup is the rather old > version > > he has. Could you elaborate with me on this? > > > -- > royj > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > From svb_fs_user at lists.adminnetz.de Tue May 7 15:42:38 2013 From: svb_fs_user at lists.adminnetz.de (svb) Date: Tue, 07 May 2013 13:42:38 +0200 Subject: [Freeswitch-users] Problem with pre-answer Message-ID: <5188E8AE.3090706@lists.adminnetz.de> Hello Freeswitch Users, I?ve got a problem with playing announcements in FreeSWITCH last stable Version 1.2.8. To be precise, when using ?session::streamFile? in a lua-script, I get the following error response: [ERR] sofia_glue.c:3015 No audio codec available The error only occurs when I set in the dialplan. On the other hand, if I do use instead, a codeclist is generated and the announcement is played. The sip-profile has the following settings: vars.xml: Furthermore the following variables are set int the dialplan: Compairing the logfiles of an ?answered? with an ?pre-answered? call I found the following Variables are NOT set in the ?pre-answered? Version: Channel-Read-Codec-Name: [PCMA] Channel-Read-Codec-Rate: [8000] Channel-Read-Codec-Bit-Rate: [64000] Channel-Write-Codec-Name: [PCMA] Channel-Write-Codec-Rate: [8000] Channel-Write-Codec-Bit-Rate: [64000] I?ve tried all the above mentioned combinations that I could think of, but without any luck. Anyone got a bright idea what actually is the matter and would kindly help me out? Regards, Sven From mogsy.uk at gmail.com Tue May 7 16:11:32 2013 From: mogsy.uk at gmail.com (Rob Moore) Date: Tue, 7 May 2013 13:11:32 +0100 Subject: [Freeswitch-users] Call Hangup cause 'None' Message-ID: Hi Guys, I was wondering if someone could clear up exactly what the hangup case 'none' means? It started appearing after an upgrade to the latest version of Freeswitch (from a very very old 1.0.0 version) in place of the usual "ALOTTED_TIMEOUT" outcomes we would expect to see from and unanswered call. I would have considered this a simple reclassification of the call result however we are still getting a few "ALOTTED_TIMEOUT" mixed in with these "none's" which makes me wonder if its actually the symptom of another problem I'm not seeing yet. Any advice on what none means and why it appears in our CDRS would be greatly appreciated! Thanks Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/c93d15bd/attachment.html From mike at jerris.com Tue May 7 16:49:17 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 7 May 2013 08:49:17 -0400 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: References: Message-ID: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> Return from a transfer? On May 7, 2013, at 8:04 AM, Grant Bagdasarian wrote: > Hello, > > When using the Originate command, is it possible to generate a dynamic sequence of actions to take place when the user answers the call? > > For example: Originate command is called with some arguments in this specific order; playback, transfer, playback. > FreeSwitch should first play something, then transfer to some destination, and after returning from the transfer, play another message. > The sequence of these actions is defined by the external application which is calling the Originate command. > > If this is possible, what is the best way to accomplish this? I have no experience with FreeSwitch, just doing some research. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/10676e15/attachment.html From bpriddy at bryantschools.org Tue May 7 16:55:25 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Tue, 7 May 2013 07:55:25 -0500 Subject: [Freeswitch-users] Can anyone out there look at this.. Message-ID: I have a pcap capture from a phone that experiences the delay. Attached is the pcap where the user had a 7 sec delay. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/3008ea05/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: delaykarla Type: application/octet-stream Size: 59379 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/3008ea05/attachment-0001.obj From ibk at labhijau.net Tue May 7 17:34:12 2013 From: ibk at labhijau.net (Iwan Budi Kusnanto) Date: Tue, 7 May 2013 20:34:12 +0700 Subject: [Freeswitch-users] How to modify the way freeswitch handle SIP OPTIONS message Message-ID: Hi, I can see from console log that freeswitch always reply with SIP 200 when receive SIP OPTIONS message. Can i modify this behaviour? How to modify it? Thanks -- Iwan Budi Kusnanto From anthony.minessale at gmail.com Tue May 7 17:48:40 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 May 2013 08:48:40 -0500 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: Is this master or stable, its not out of the question the param has a bug in master since there were some major changes and some media related params may need testing. The easy way would be lab it up and file a Jira so it can be quantified and resolved. On May 7, 2013 7:28 AM, "Steven Ayre" wrote: > No, it should actually hang up the call... > > Steve > > > > On 7 May 2013, at 10:12, Farrukh Ali wrote: > > Hey Mehroz, > > The problem with this timer is it only *STOPS *the RTP traffic going from > FS to SIPClient by detecting RTP is not received from the SIPClient for > certain time(you set), It doesn't kill the session as you can check it by > pressing F4, you need to hupall all the time, In Wireshark you could see > that RTP stream is stopped after the timer expires! > > Regards > Muhammad Farrukh > > > > > Date: Mon, 6 May 2013 23:15:18 -0700 > > From: mehroz.ashraf85 at gmail.com > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > > > I have already got that at the right place Farrukh! > > > > > > > > > > > > > > > > But unable to get the desired result! calls does not hangs up , which i > > monitored for 5 min and have to do "hupall" to get it fixed. > > > > Is this parameter have any dependency over any other parameter , liker > > timers? as i have disabled session timer > > param name="enable-timer" value="false", but AFAIK session timer and > > rtp-timers are two different entities! > > > > > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.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://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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/8021ff92/attachment.html From GB at cm.nl Tue May 7 17:52:22 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Tue, 7 May 2013 15:52:22 +0200 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> Message-ID: Yes. For example: - Originate to A. - Playback "Hello World" - Transfer A to B. - B hangs up the call. - Continue with call flow. Playback "Bye". This is what I've done many times with VoiceXML. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, May 7, 2013 2:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Generate dynamic dialplan based on variables Return from a transfer? On May 7, 2013, at 8:04 AM, Grant Bagdasarian > wrote: Hello, When using the Originate command, is it possible to generate a dynamic sequence of actions to take place when the user answers the call? For example: Originate command is called with some arguments in this specific order; playback, transfer, playback. FreeSwitch should first play something, then transfer to some destination, and after returning from the transfer, play another message. The sequence of these actions is defined by the external application which is calling the Originate command. If this is possible, what is the best way to accomplish this? I have no experience with FreeSwitch, just doing some research. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/72e0953b/attachment.html From bdfoster at davri.com Tue May 7 18:05:49 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 7 May 2013 10:05:49 -0400 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> Message-ID: Isnt that the point behind using a dialplan? On May 7, 2013 9:57 AM, "Grant Bagdasarian" wrote: > Yes. For example: **** > > **- **Originate to A.**** > > **- **Playback ?Hello World?**** > > **- **Transfer A to B.**** > > **- **B hangs up the call.**** > > **- **Continue with call flow. Playback ?Bye?.**** > > ** ** > > This is what I?ve done many times with VoiceXML.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Jerris > *Sent:* Tuesday, May 7, 2013 2:49 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Generate dynamic dialplan based on > variables**** > > ** ** > > Return from a transfer?**** > > ** ** > > On May 7, 2013, at 8:04 AM, Grant Bagdasarian wrote:**** > > > > **** > > Hello,**** > > **** > > When using the Originate command, is it possible to generate a dynamic > sequence of actions to take place when the user answers the call?**** > > **** > > For example: Originate command is called with some arguments in this > specific order; playback, transfer, playback.**** > > FreeSwitch should first play something, then transfer to some destination, > and after returning from the transfer, play another message.**** > > The sequence of these actions is defined by the external application which > is calling the Originate command.**** > > **** > > If this is possible, what is the best way to accomplish this? I have no > experience with FreeSwitch, just doing some research.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/c647a5e5/attachment-0001.html From mike at jerris.com Tue May 7 18:15:20 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 7 May 2013 10:15:20 -0400 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> Message-ID: <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> This isn't a transfer, its just bridging to B. Yes, this can be done with dialplan, make sure to set hangup_after_bridge=false so that it can continue after the transfer. you can use originate to an extension with those applications in it or use inline dialplan with all the actions. Mike On May 7, 2013, at 9:52 AM, Grant Bagdasarian wrote: > Yes. For example: > - Originate to A. > - Playback ?Hello World? > - Transfer A to B. > - B hangs up the call. > - Continue with call flow. Playback ?Bye?. > > This is what I?ve done many times with VoiceXML. > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris > Sent: Tuesday, May 7, 2013 2:49 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Generate dynamic dialplan based on variables > > Return from a transfer? > > On May 7, 2013, at 8:04 AM, Grant Bagdasarian wrote: > > > Hello, > > When using the Originate command, is it possible to generate a dynamic sequence of actions to take place when the user answers the call? > > For example: Originate command is called with some arguments in this specific order; playback, transfer, playback. > FreeSwitch should first play something, then transfer to some destination, and after returning from the transfer, play another message. > The sequence of these actions is defined by the external application which is calling the Originate command. > > If this is possible, what is the best way to accomplish this? I have no experience with FreeSwitch, just doing some research. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/a4c76fa2/attachment.html From mbodbg at gmx.net Tue May 7 19:06:54 2013 From: mbodbg at gmx.net (mbo) Date: Tue, 7 May 2013 17:06:54 +0200 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: <7379ED61-0909-46C5-ACFA-C502DC9C78A3@gmx.net> We did some more tests an found out that if we add a Event-UUID to the sendmsg command like: sendmsg Event-UUID: 5bf340cd-7a7e-4965-9285-95ed365ed242 call-command: execute execute-app-name: speak execute-app-arg: flite|slt|One We will get back this Event-UUID as Application-UUID in the CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE events, so we are able to match it. In our tests this works fine, however I'm not sure if this approach is reliable and has any side effects. May I ask for your opinion of this approach? Thanks Markus Am 05.05.2013 um 20:58 schrieb mbo : > Thanks for the answers. There is hardly any documentation of the PRIVATE_COMMAND event, can you give me some more details on it. What is it good for, when is it send, etc? > > If it is just an additional event including the channel data, where is the benefit using this event instead of set a custom variable first and then check this variable in the ChannelExecuteComplete event? Or am I missing something and it is possible to set a custom channel variable during SendMsg? > > Thanks > > Markus > > > Am 03.05.2013 um 21:13 schrieb Anthony Minessale : > >> Compromise: >> >> Update to latest and sub to the "private_command" event, you should get back you own event with channel data merged in. >> >> >> >> On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: >> >> >> On Fri, May 3, 2013 at 2:19 AM, mbo wrote: >> I'm referring to a two years old bug report http://jira.freeswitch.org/browse/FS-1309. >> >> Is it in the meantime somehow possible to map reply to SendMsg in asyc mode? I'm wondering why this ticket has been closed as "Won't fix", in my opinion is an essential feature to handle events properly. >> >> If not, I want to implement the work around described in the ticket, to set a channel variable in a round trip before executing the real command. Do I need to wait for the ChannelExecuteComplete event for the Set command, or can I send my "real" command right away after the Set command? >> To be safe you should verify that the set app actually ran before you implement the workaround. >> -MC >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/0da1c7d2/attachment-0001.html From anthony.minessale at gmail.com Tue May 7 19:35:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 May 2013 10:35:11 -0500 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: This is the first time the event is used. I changed to code in real time for you based on your request so you will be the one doing the documentation. That's how open source works. The point once you update and compile master, your actual sendmsg event that you send will now be sent back to you once it is processed with every header you supplied as well as the caller profile data from the channel you sent it on. This was what you asked for. On Sun, May 5, 2013 at 1:58 PM, mbo wrote: > Thanks for the answers. There is hardly any documentation of the > PRIVATE_COMMAND event, can you give me some more details on it. What is it > good for, when is it send, etc? > > If it is just an additional event including the channel data, where is the > benefit using this event instead of set a custom variable first and then > check this variable in the ChannelExecuteComplete event? Or am I missing > something and it is possible to set a custom channel variable during > SendMsg? > > Thanks > > Markus > > > Am 03.05.2013 um 21:13 schrieb Anthony Minessale < > anthony.minessale at gmail.com>: > > Compromise: > > Update to latest and sub to the "private_command" event, you should get > back you own event with channel data merged in. > > > > On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: > >> >> >> On Fri, May 3, 2013 at 2:19 AM, mbo wrote: >> >>> I'm referring to a two years old bug report >>> http://jira.freeswitch.org/browse/FS-1309. >>> >>> Is it in the meantime somehow possible to map reply to SendMsg in asyc >>> mode? I'm wondering why this ticket has been closed as "Won't fix", in my >>> opinion is an essential feature to handle events properly. >>> >>> If not, I want to implement the work around described in the ticket, to >>> set a channel variable in a round trip before executing the real command. >>> Do I need to wait for the ChannelExecuteComplete event for the Set command, >>> or can I send my "real" command right away after the Set command? >>> >> To be safe you should verify that the set app actually ran before you >> implement the workaround. >> -MC >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/d3f2c7f2/attachment.html From spencer at 5ninesolutions.com Tue May 7 20:03:00 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 7 May 2013 09:03:00 -0700 Subject: [Freeswitch-users] Codec Negotiation Help Message-ID: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> Hello all, I'm having a bit of trouble getting codec negotiation working the way I'd like to. I'm trying to avoid transcoding unless its absolutely necessary for the call to be setup. I use two profiles with late negotiation enabled, an internal facing registered user endpoints and an external facing the PSTN via various ITSPs. For calls to users I've been setting codec_string=${ep_codec_string} in my dial strings and normal bridges work fine but I'm now reworking the hunt groups using enterprise originate and that doesn't seem to work correctly (the callee's preferences are always used). So I'd like to do the the "right" way now :-) The basic scenario is this: On the internal profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM On the external profile I have: CODECS IN G722,PCMU,G729 CODECS OUT G722,PCMU,G729 What I'm trying to accomplish is this: Call from PSTN, use caller's codecs as priority for the bridge, i.e. PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local Endpoint PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local Endpoint PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local Endpoint Call from local endpoint to PSTN, use callee's codecs as priority for the bridge, i.e. Local Endpoint Offers G722, PCMU, G729 GW Responds PCMU,G729 Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> PSTN Desired negotiation: Both legs use PCMU Local Endpoint Offers G722, PCMU, G729 GW Responds G729,PCMU Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Both legs use G729 Local Endpoint Offers GSM GW Responds PCMU, G729 Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH transcodes. Any help is greatly appreciated! Thanks, Spencer From anthony.minessale at gmail.com Tue May 7 20:20:11 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 May 2013 11:20:11 -0500 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: too slow, I found the problem in master, if you don't want to update to get the fix, use the variable version of the param instead by setting rtp_timeout_sec in vars.xml On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Is this master or stable, its not out of the question the param has a bug > in master since there were some major changes and some media related params > may need testing. The easy way would be lab it up and file a Jira so it > can be quantified and resolved. > On May 7, 2013 7:28 AM, "Steven Ayre" wrote: > >> No, it should actually hang up the call... >> >> Steve >> >> >> >> On 7 May 2013, at 10:12, Farrukh Ali wrote: >> >> Hey Mehroz, >> >> The problem with this timer is it only *STOPS *the RTP traffic going >> from FS to SIPClient by detecting RTP is not received from the SIPClient >> for certain time(you set), It doesn't kill the session as you can check it >> by pressing F4, you need to hupall all the time, In Wireshark you could see >> that RTP stream is stopped after the timer expires! >> >> Regards >> Muhammad Farrukh >> >> >> >> > Date: Mon, 6 May 2013 23:15:18 -0700 >> > From: mehroz.ashraf85 at gmail.com >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: Re: [Freeswitch-users] Dead channels in freeswitch >> > >> > I have already got that at the right place Farrukh! >> > >> > >> > >> > >> > >> > >> > >> > But unable to get the desired result! calls does not hangs up , which i >> > monitored for 5 min and have to do "hupall" to get it fixed. >> > >> > Is this parameter have any dependency over any other parameter , liker >> > timers? as i have disabled session timer >> > param name="enable-timer" value="false", but AFAIK session timer and >> > rtp-timers are two different entities! >> > >> > >> > >> > -- >> > View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.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://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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/ae8b13dd/attachment-0001.html From mbodbg at gmx.net Tue May 7 21:02:25 2013 From: mbodbg at gmx.net (mbo) Date: Tue, 7 May 2013 19:02:25 +0200 Subject: [Freeswitch-users] channel selection during outbound dialing Message-ID: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in one group. In Asterisk it is possible to specify the channel selection mode by a prefix in the dial command (Dial(Zap/g2?) - ascending sequential hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2?) - descending sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I use a dial string like: freetdm/mygroup/a/ to dial out via a group. I'm not sure for what the "a" stand for. Are there also options for the channel selection mode available like in asterisk? Thanks Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/e487746a/attachment.html From steveayre at gmail.com Tue May 7 22:14:47 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 7 May 2013 19:14:47 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: Guessing this is related to the move of the media handling to crtp? -Steve On 7 May 2013 17:20, Anthony Minessale wrote: > too slow, > > I found the problem in master, if you don't want to update to get the fix, > use the variable version of the param instead by setting rtp_timeout_sec in > vars.xml > > > > > On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale > wrote: >> >> Is this master or stable, its not out of the question the param has a bug >> in master since there were some major changes and some media related params >> may need testing. The easy way would be lab it up and file a Jira so it can >> be quantified and resolved. >> >> On May 7, 2013 7:28 AM, "Steven Ayre" wrote: >>> >>> No, it should actually hang up the call... >>> >>> Steve >>> >>> >>> >>> On 7 May 2013, at 10:12, Farrukh Ali wrote: >>> >>> Hey Mehroz, >>> >>> The problem with this timer is it only STOPS the RTP traffic going from >>> FS to SIPClient by detecting RTP is not received from the SIPClient for >>> certain time(you set), It doesn't kill the session as you can check it by >>> pressing F4, you need to hupall all the time, In Wireshark you could see >>> that RTP stream is stopped after the timer expires! >>> >>> Regards >>> Muhammad Farrukh >>> >>> >>> >>> > Date: Mon, 6 May 2013 23:15:18 -0700 >>> > From: mehroz.ashraf85 at gmail.com >>> > To: freeswitch-users at lists.freeswitch.org >>> > Subject: Re: [Freeswitch-users] Dead channels in freeswitch >>> > >>> > I have already got that at the right place Farrukh! >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > But unable to get the desired result! calls does not hangs up , which i >>> > monitored for 5 min and have to do "hupall" to get it fixed. >>> > >>> > Is this parameter have any dependency over any other parameter , liker >>> > timers? as i have disabled session timer >>> > param name="enable-timer" value="false", but AFAIK session timer and >>> > rtp-timers are two different entities! >>> > >>> > >>> > >>> > -- >>> > View this message in context: >>> > http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590411.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://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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From steveayre at gmail.com Tue May 7 22:17:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 7 May 2013 19:17:37 +0100 Subject: [Freeswitch-users] How to modify the way freeswitch handle SIP OPTIONS message In-Reply-To: References: Message-ID: That's the correct behaviour... what behaviour are you after? There's also a sofia profile parameter sip-options-respond-503-on-busy. At the moment that's the only other option. -Steve On 7 May 2013 14:34, Iwan Budi Kusnanto wrote: > Hi, > I can see from console log that freeswitch always reply with SIP 200 > when receive SIP OPTIONS message. > Can i modify this behaviour? How to modify it? > > Thanks > > -- > Iwan Budi Kusnanto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mike at jerris.com Tue May 7 22:28:25 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 7 May 2013 14:28:25 -0400 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> Message-ID: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> what is crtp? On May 7, 2013, at 2:14 PM, Steven Ayre wrote: > Guessing this is related to the move of the media handling to crtp? > > -Steve > > On 7 May 2013 17:20, Anthony Minessale wrote: >> too slow, >> >> I found the problem in master, if you don't want to update to get the fix, >> use the variable version of the param instead by setting rtp_timeout_sec in >> vars.xml >> >> On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale >> wrote: >>> >>> Is this master or stable, its not out of the question the param has a bug >>> in master since there were some major changes and some media related params >>> may need testing. The easy way would be lab it up and file a Jira so it can >>> be quantified and resolved. >>> >>> On May 7, 2013 7:28 AM, "Steven Ayre" wrote: >>>> >>>> No, it should actually hang up the call... >>>> >>>> Steve >>>> >>>> On 7 May 2013, at 10:12, Farrukh Ali wrote: >>>> >>>> Hey Mehroz, >>>> >>>> The problem with this timer is it only STOPS the RTP traffic going from >>>> FS to SIPClient by detecting RTP is not received from the SIPClient for >>>> certain time(you set), It doesn't kill the session as you can check it by >>>> pressing F4, you need to hupall all the time, In Wireshark you could see >>>> that RTP stream is stopped after the timer expires! >>>> >>>> Regards >>>> Muhammad Farrukh >>>>> Date: Mon, 6 May 2013 23:15:18 -0700 >>>>> From: mehroz.ashraf85 at gmail.com >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> Subject: Re: [Freeswitch-users] Dead channels in freeswitch >>>>> >>>>> I have already got that at the right place Farrukh! >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> But unable to get the desired result! calls does not hangs up , which i >>>>> monitored for 5 min and have to do "hupall" to get it fixed. >>>>> >>>>> Is this parameter have any dependency over any other parameter , liker >>>>> timers? as i have disabled session timer >>>>> param name="enable-timer" value="false", but AFAIK session timer and >>>>> rtp-timers are two different entities! >>>>> From steveayre at gmail.com Tue May 7 23:11:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 7 May 2013 20:11:39 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> References: <1367907318527-7590411.post@n2.nabble.com> <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> Message-ID: commit ef5c1256f3a64502880bb232ea17a673e3e4eb10 Author: Anthony Minessale Date: Wed Aug 8 14:46:38 2012 -0500 add rtp endpoint contributed by sangoma I'm guessing it's called crtp since there's a crtp_init function. -Steve On 7 May 2013 19:28, Michael Jerris wrote: > what is crtp? > > On May 7, 2013, at 2:14 PM, Steven Ayre wrote: > >> Guessing this is related to the move of the media handling to crtp? >> >> -Steve >> >> On 7 May 2013 17:20, Anthony Minessale wrote: >>> too slow, >>> >>> I found the problem in master, if you don't want to update to get the fix, >>> use the variable version of the param instead by setting rtp_timeout_sec in >>> vars.xml >>> >>> On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale >>> wrote: >>>> >>>> Is this master or stable, its not out of the question the param has a bug >>>> in master since there were some major changes and some media related params >>>> may need testing. The easy way would be lab it up and file a Jira so it can >>>> be quantified and resolved. >>>> >>>> On May 7, 2013 7:28 AM, "Steven Ayre" wrote: >>>>> >>>>> No, it should actually hang up the call... >>>>> >>>>> Steve >>>>> >>>>> On 7 May 2013, at 10:12, Farrukh Ali wrote: >>>>> >>>>> Hey Mehroz, >>>>> >>>>> The problem with this timer is it only STOPS the RTP traffic going from >>>>> FS to SIPClient by detecting RTP is not received from the SIPClient for >>>>> certain time(you set), It doesn't kill the session as you can check it by >>>>> pressing F4, you need to hupall all the time, In Wireshark you could see >>>>> that RTP stream is stopped after the timer expires! >>>>> >>>>> Regards >>>>> Muhammad Farrukh >>>>>> Date: Mon, 6 May 2013 23:15:18 -0700 >>>>>> From: mehroz.ashraf85 at gmail.com >>>>>> To: freeswitch-users at lists.freeswitch.org >>>>>> Subject: Re: [Freeswitch-users] Dead channels in freeswitch >>>>>> >>>>>> I have already got that at the right place Farrukh! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> But unable to get the desired result! calls does not hangs up , which i >>>>>> monitored for 5 min and have to do "hupall" to get it fixed. >>>>>> >>>>>> Is this parameter have any dependency over any other parameter , liker >>>>>> timers? as i have disabled session timer >>>>>> param name="enable-timer" value="false", but AFAIK session timer and >>>>>> rtp-timers are two different entities! >>>>>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From victor.chukalovskiy at gmail.com Tue May 7 23:18:15 2013 From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy) Date: Tue, 07 May 2013 15:18:15 -0400 Subject: [Freeswitch-users] No ring-back with 180 ringing and P-Early-Media tag Message-ID: <51895377.8090109@gmail.com> Hello, Can FreeSWITCH be configured not to include P-Early-Media tag in it's 180 ringing? Is any of the boxes involved breaking the specs? Call scenario is that far-end SBC sends INVITE to FS with /P-Early-Media: supported/ tag. Freeswitch routes call and replies with 180 Ringing with /P-Early-Media: sendonly/ tag to the far-end SBC. Far-end does not generate ringback when it sees P-Early-Media: sendonly. Freswitch does not provide in-band ring-back either (cause it's just 180 Ringing without SDP). Last but not least, FreeSWITCH runs in bypass media mode. Thank you, Victor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/74472276/attachment-0001.html From anthony.minessale at gmail.com Tue May 7 23:23:41 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 7 May 2013 14:23:41 -0500 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> Message-ID: Nope, that's just an endpoint that has no signalling for use in some sangoma product. Its still our same rtp stack. On Tue, May 7, 2013 at 2:11 PM, Steven Ayre wrote: > commit ef5c1256f3a64502880bb232ea17a673e3e4eb10 > Author: Anthony Minessale > Date: Wed Aug 8 14:46:38 2012 -0500 > add rtp endpoint contributed by sangoma > > I'm guessing it's called crtp since there's a crtp_init function. > > -Steve > > > > > On 7 May 2013 19:28, Michael Jerris wrote: > > what is crtp? > > > > On May 7, 2013, at 2:14 PM, Steven Ayre wrote: > > > >> Guessing this is related to the move of the media handling to crtp? > >> > >> -Steve > >> > >> On 7 May 2013 17:20, Anthony Minessale > wrote: > >>> too slow, > >>> > >>> I found the problem in master, if you don't want to update to get the > fix, > >>> use the variable version of the param instead by setting > rtp_timeout_sec in > >>> vars.xml > >>> > >>> On Tue, May 7, 2013 at 8:48 AM, Anthony Minessale > >>> wrote: > >>>> > >>>> Is this master or stable, its not out of the question the param has a > bug > >>>> in master since there were some major changes and some media related > params > >>>> may need testing. The easy way would be lab it up and file a Jira so > it can > >>>> be quantified and resolved. > >>>> > >>>> On May 7, 2013 7:28 AM, "Steven Ayre" wrote: > >>>>> > >>>>> No, it should actually hang up the call... > >>>>> > >>>>> Steve > >>>>> > >>>>> On 7 May 2013, at 10:12, Farrukh Ali wrote: > >>>>> > >>>>> Hey Mehroz, > >>>>> > >>>>> The problem with this timer is it only STOPS the RTP traffic going > from > >>>>> FS to SIPClient by detecting RTP is not received from the SIPClient > for > >>>>> certain time(you set), It doesn't kill the session as you can check > it by > >>>>> pressing F4, you need to hupall all the time, In Wireshark you could > see > >>>>> that RTP stream is stopped after the timer expires! > >>>>> > >>>>> Regards > >>>>> Muhammad Farrukh > >>>>>> Date: Mon, 6 May 2013 23:15:18 -0700 > >>>>>> From: mehroz.ashraf85 at gmail.com > >>>>>> To: freeswitch-users at lists.freeswitch.org > >>>>>> Subject: Re: [Freeswitch-users] Dead channels in freeswitch > >>>>>> > >>>>>> I have already got that at the right place Farrukh! > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> But unable to get the desired result! calls does not hangs up , > which i > >>>>>> monitored for 5 min and have to do "hupall" to get it fixed. > >>>>>> > >>>>>> Is this parameter have any dependency over any other parameter , > liker > >>>>>> timers? as i have disabled session timer > >>>>>> param name="enable-timer" value="false", but AFAIK session timer and > >>>>>> rtp-timers are two different entities! > >>>>>> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/0e0f909b/attachment.html From msc at freeswitch.org Tue May 7 23:28:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 12:28:53 -0700 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> Message-ID: And if you want it to be extremely dynamic you can send the originated leg into an xml_curl-generated dialplan or use mod_httapi to create something on the fly. -MC On Tue, May 7, 2013 at 7:15 AM, Michael Jerris wrote: > This isn't a transfer, its just bridging to B. > > Yes, this can be done with dialplan, make sure to set > hangup_after_bridge=false so that it can continue after the transfer. you > can use originate to an extension with those applications in it or use > inline dialplan with all the actions. > > Mike > > > On May 7, 2013, at 9:52 AM, Grant Bagdasarian wrote: > > Yes. For example:**** > - Originate to A.**** > - Playback ?Hello World?**** > - Transfer A to B.**** > - B hangs up the call.**** > - Continue with call flow. Playback ?Bye?.**** > > This is what I?ve done many times with VoiceXML.**** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Jerris > *Sent:* Tuesday, May 7, 2013 2:49 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Generate dynamic dialplan based on > variables**** > ** ** > Return from a transfer?**** > ** ** > On May 7, 2013, at 8:04 AM, Grant Bagdasarian wrote:**** > > > **** > Hello,**** > **** > When using the Originate command, is it possible to generate a dynamic > sequence of actions to take place when the user answers the call?**** > **** > For example: Originate command is called with some arguments in this > specific order; playback, transfer, playback.**** > FreeSwitch should first play something, then transfer to some destination, > and after returning from the transfer, play another message.**** > The sequence of these actions is defined by the external application which > is calling the Originate command.**** > **** > If this is possible, what is the best way to accomplish this? I have no > experience with FreeSwitch, just doing some research.**** > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/998c47f8/attachment-0001.html From msc at freeswitch.org Tue May 7 23:41:46 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 12:41:46 -0700 Subject: [Freeswitch-users] channel selection during outbound dialing In-Reply-To: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> References: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> Message-ID: I believe all of that is covered here: http://wiki.freeswitch.org/wiki/FreeTDM#Dial_Plan The "a" means start at channel 1 and work your way up. -MC On Tue, May 7, 2013 at 10:02 AM, mbo wrote: > We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in one > group. In Asterisk it is possible to specify the channel selection mode > by a prefix in the dial command (Dial(Zap/g2?) - ascending sequential > hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2?) - descending > sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I > use a dial string like: > > freetdm/mygroup/a/ > > to dial out via a group. I'm not sure for what the "a" stand for. Are > there also options for the channel selection mode available like in > asterisk? > > Thanks > > Markus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/18e39d1b/attachment.html From msc at freeswitch.org Tue May 7 23:58:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 12:58:22 -0700 Subject: [Freeswitch-users] settings for a Trixbox v 2.6 connection to a Sip Trunking translate to gateway using Freeswitch 1.2.0 In-Reply-To: References: Message-ID: Richard, Did you figure this out yet? I suspect that your gateway would look something like this: Note that FreeSWITCH does not transcode g729 without a license so be sure to buy one at FreeSWITCH.org if you intend to use g729. If not you can probably just use PCMU (i.e. g711u) with the carrier. We don't have a Tricom entry in our provider configuration pagesso if you get it working we'd appreciate it if you'd let us know. -MC On Fri, May 3, 2013 at 2:26 PM, Richard Cruz wrote: > Hi There! > > I have the following settings for a Trixbox v 2.6 connection to a Sip > Trunking at tricom.net > > Host=XXX.X.132.37 > defaultuser=********** > Secret= *********** > Fromuser=xxxxxxxxxx > Type=friend > insecure=invite,port > Dtmfmode=rfc2833 > Disallow=all > Allow=g729 > Canredirect=no > Canreivite=no > Context=from-trunk > Qualify=yes > > > and I want to translate to gateway using Freeswitch 1.2.0 > -- > Richard Cruz > 678.394-6400 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/0ed20fb2/attachment.html From richardrcruzc at gmail.com Wed May 8 00:14:37 2013 From: richardrcruzc at gmail.com (Richard Cruz) Date: Tue, 7 May 2013 16:14:37 -0400 Subject: [Freeswitch-users] settings for a Trixbox v 2.6 connection to a Sip Trunking translate to gateway using Freeswitch 1.2.0 In-Reply-To: References: Message-ID: Michael, yes I did figured out that... this is how it's working now... but i till have some issue with codec g729 to tricom, i using xlite free version. On Tue, May 7, 2013 at 3:58 PM, Michael Collins wrote: > Richard, > > Did you figure this out yet? I suspect that your gateway would look > something like this: > > > > > > > > > Note that FreeSWITCH does not transcode g729 without a license so be sure > to buy one at FreeSWITCH.org if you intend to use g729. If not you can > probably just use PCMU (i.e. g711u) with the carrier. > > We don't have a Tricom entry in our provider configuration pagesso if you get it working we'd appreciate it if you'd let us know. > > -MC > > On Fri, May 3, 2013 at 2:26 PM, Richard Cruz wrote: > >> Hi There! >> >> I have the following settings for a Trixbox v 2.6 connection to a Sip >> Trunking at tricom.net >> >> Host=XXX.X.132.37 >> defaultuser=********** >> Secret= *********** >> Fromuser=xxxxxxxxxx >> Type=friend >> insecure=invite,port >> Dtmfmode=rfc2833 >> Disallow=all >> Allow=g729 >> Canredirect=no >> Canreivite=no >> Context=from-trunk >> Qualify=yes >> >> >> and I want to translate to gateway using Freeswitch 1.2.0 >> -- >> Richard Cruz >> 678.394-6400 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Richard Cruz 678.394-6400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/6729ed66/attachment-0001.html From msc at freeswitch.org Wed May 8 00:27:18 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 13:27:18 -0700 Subject: [Freeswitch-users] settings for a Trixbox v 2.6 connection to a Sip Trunking translate to gateway using Freeswitch 1.2.0 In-Reply-To: References: Message-ID: If you want to use g729 then you'll need to set bypass_media=true in your dialplan. More information is available on the wiki. Also, you should probably join #freeswitch on irc.freenode.net so that you can talk to people in realtime. -MC On Tue, May 7, 2013 at 1:14 PM, Richard Cruz wrote: > Michael, yes I did figured out that... > this is how it's working now... but i till have some issue with codec g729 > to tricom, i using xlite free version. > > > > > > > > > > data="effective_caller_id_number=809475xxxx"/> > > > > > > > On Tue, May 7, 2013 at 3:58 PM, Michael Collins wrote: > >> Richard, >> >> Did you figure this out yet? I suspect that your gateway would look >> something like this: >> >> >> >> >> >> >> >> >> Note that FreeSWITCH does not transcode g729 without a license so be sure >> to buy one at FreeSWITCH.org if you intend to use g729. If not you can >> probably just use PCMU (i.e. g711u) with the carrier. >> >> We don't have a Tricom entry in our provider configuration pagesso if you get it working we'd appreciate it if you'd let us know. >> >> -MC >> >> On Fri, May 3, 2013 at 2:26 PM, Richard Cruz wrote: >> >>> Hi There! >>> >>> I have the following settings for a Trixbox v 2.6 connection to a Sip >>> Trunking at tricom.net >>> >>> Host=XXX.X.132.37 >>> defaultuser=********** >>> Secret= *********** >>> Fromuser=xxxxxxxxxx >>> Type=friend >>> insecure=invite,port >>> Dtmfmode=rfc2833 >>> Disallow=all >>> Allow=g729 >>> Canredirect=no >>> Canreivite=no >>> Context=from-trunk >>> Qualify=yes >>> >>> >>> and I want to translate to gateway using Freeswitch 1.2.0 >>> -- >>> Richard Cruz >>> 678.394-6400 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Richard Cruz > 678.394-6400 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/cbcd5cdc/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed May 8 00:28:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 7 May 2013 21:28:15 +0100 Subject: [Freeswitch-users] Request for review - FS-5388 Message-ID: Hello! Could a core dev spend a minute or two reviewing the following ticket please; http://jira.freeswitch.org/browse/FS-5388 In return, I have triaged the following ticket; http://jira.freeswitch.org/browse/FS-5168 I spent 10 minutes looking for other tickets I could help with, but all of them were beyond my skill level :/ Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/35291803/attachment.html From msc at freeswitch.org Wed May 8 00:30:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 13:30:54 -0700 Subject: [Freeswitch-users] Codec Negotiation Help In-Reply-To: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> References: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> Message-ID: This may just be not setting the chan variable correctly in your enterprise originate string. What does your enterprise originate look like? -MC On Tue, May 7, 2013 at 9:03 AM, Spencer Thomason wrote: > Hello all, > I'm having a bit of trouble getting codec negotiation working the way I'd > like to. I'm trying to avoid transcoding unless its absolutely necessary > for the call to be setup. I use two profiles with late negotiation > enabled, an internal facing registered user endpoints and an external > facing the PSTN via various ITSPs. For calls to users I've been setting > codec_string=${ep_codec_string} in my dial strings and normal bridges work > fine but I'm now reworking the hunt groups using enterprise originate and > that doesn't seem to work correctly (the callee's preferences are always > used). So I'd like to do the the "right" way now :-) > > The basic scenario is this: > > On the internal profile I have: > CODECS IN G722,PCMU,G729,GSM > CODECS OUT G722,PCMU,G729,GSM > > On the external profile I have: > CODECS IN G722,PCMU,G729 > CODECS OUT G722,PCMU,G729 > > What I'm trying to accomplish is this: > Call from PSTN, use caller's codecs as priority for the bridge, i.e. > PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local > Endpoint > PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local > Endpoint > PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local > Endpoint > > > Call from local endpoint to PSTN, use callee's codecs as priority for the > bridge, i.e. > Local Endpoint Offers G722, PCMU, G729 > GW Responds PCMU,G729 > Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> > PSTN > Desired negotiation: Both legs use PCMU > > Local Endpoint Offers G722, PCMU, G729 > GW Responds G729,PCMU > Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> > PSTN > Desired negotiation: Both legs use G729 > > Local Endpoint Offers GSM > GW Responds PCMU, G729 > Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN > Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH > transcodes. > > Any help is greatly appreciated! > > > Thanks, > Spencer > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/b7d8316f/attachment-0001.html From robert.hadley at teotech.com Wed May 8 00:37:35 2013 From: robert.hadley at teotech.com (Robert Hadley) Date: Tue, 7 May 2013 20:37:35 +0000 Subject: [Freeswitch-users] Codec Negotiation Help In-Reply-To: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> References: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> Message-ID: <6f28692a07874746b81063da30587524@BLUPR04MB024.namprd04.prod.outlook.com> Hi Spencer, I also use late_negotiation and inherit_codec=true. I configure outbound codecs to append ep_codec_string using originator_codec. Regards, Robert -----Original Message----- From: Spencer Thomason [mailto:spencer at 5ninesolutions.com] Sent: Tuesday, May 07, 2013 9:03 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Codec Negotiation Help Hello all, I'm having a bit of trouble getting codec negotiation working the way I'd like to. I'm trying to avoid transcoding unless its absolutely necessary for the call to be setup. I use two profiles with late negotiation enabled, an internal facing registered user endpoints and an external facing the PSTN via various ITSPs. For calls to users I've been setting codec_string=${ep_codec_string} in my dial strings and normal bridges work fine but I'm now reworking the hunt groups using enterprise originate and that doesn't seem to work correctly (the callee's preferences are always used). So I'd like to do the the "right" way now :-) The basic scenario is this: On the internal profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM On the external profile I have: CODECS IN G722,PCMU,G729 CODECS OUT G722,PCMU,G729 What I'm trying to accomplish is this: Call from PSTN, use caller's codecs as priority for the bridge, i.e. PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local Endpoint PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local Endpoint PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local Endpoint Call from local endpoint to PSTN, use callee's codecs as priority for the bridge, i.e. Local Endpoint Offers G722, PCMU, G729 GW Responds PCMU,G729 Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> PSTN Desired negotiation: Both legs use PCMU Local Endpoint Offers G722, PCMU, G729 GW Responds G729,PCMU Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Both legs use G729 Local Endpoint Offers GSM GW Responds PCMU, G729 Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH transcodes. Any help is greatly appreciated! Thanks, Spencer From jmesquita at freeswitch.org Wed May 8 01:40:14 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 7 May 2013 18:40:14 -0300 Subject: [Freeswitch-users] Request for review - FS-5388 In-Reply-To: References: Message-ID: Cal, that tends to be my feeling as well when going trhu Jira but I can't explain how valuable it has been to simply follow them. It helps a lot if you also try to reproduce Jira's as they show up or the reporter has disappeared... Or better yet, help us poke people. I do that constantly and it is hard to get feedback sometimes. Jo?o Mesquita FreeSWITCH? Solutions On Tue, May 7, 2013 at 5:28 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello! > > Could a core dev spend a minute or two reviewing the following ticket > please; > http://jira.freeswitch.org/browse/FS-5388 > > In return, I have triaged the following ticket; > http://jira.freeswitch.org/browse/FS-5168 > > I spent 10 minutes looking for other tickets I could help with, but all of > them were beyond my skill level :/ > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/f4762551/attachment.html From intralanman at freeswitch.org Wed May 8 01:43:16 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 07 May 2013 17:43:16 -0400 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> Message-ID: <51897574.9080705@freeswitch.org> On 13-05-07 03:28 PM, Michael Collins wrote: > And if you want it to be extremely dynamic you can send the originated > leg into an xml_curl-generated dialplan or use mod_httapi to create > something on the fly. I think that's covered in the new book that's coming out soon. That might be a worthwhile investment to make ;-) -Ray From msc at freeswitch.org Wed May 8 01:47:48 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 May 2013 14:47:48 -0700 Subject: [Freeswitch-users] Request for review - FS-5388 In-Reply-To: References: Message-ID: On Tue, May 7, 2013 at 2:40 PM, Jo?o Mesquita wrote: > Cal, that tends to be my feeling as well when going trhu Jira but I can't > explain how valuable it has been to simply follow them. It helps a lot if > you also try to reproduce Jira's as they show up or the reporter has > disappeared... Or better yet, help us poke people. I do that constantly and > it is hard to get feedback sometimes. > > Jo?o Mesquita > FreeSWITCH? Solutions > We thank you gentlemen for your diligent efforts! Please keep up the good work. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/8ebf7feb/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed May 8 01:51:11 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 7 May 2013 22:51:11 +0100 Subject: [Freeswitch-users] Request for review - FS-5388 In-Reply-To: References: Message-ID: Got it - I'll dig a bit deeper next time and do some chasing where possible! Cal On Tue, May 7, 2013 at 10:40 PM, Jo?o Mesquita wrote: > Cal, that tends to be my feeling as well when going trhu Jira but I can't > explain how valuable it has been to simply follow them. It helps a lot if > you also try to reproduce Jira's as they show up or the reporter has > disappeared... Or better yet, help us poke people. I do that constantly and > it is hard to get feedback sometimes. > > Jo?o Mesquita > FreeSWITCH? Solutions > > > On Tue, May 7, 2013 at 5:28 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello! >> >> Could a core dev spend a minute or two reviewing the following ticket >> please; >> http://jira.freeswitch.org/browse/FS-5388 >> >> In return, I have triaged the following ticket; >> http://jira.freeswitch.org/browse/FS-5168 >> >> I spent 10 minutes looking for other tickets I could help with, but all >> of them were beyond my skill level :/ >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130507/f9d81f17/attachment-0001.html From muhdnordin at gmail.com Wed May 8 02:31:07 2013 From: muhdnordin at gmail.com (nordin mohamed) Date: Wed, 8 May 2013 06:31:07 +0800 Subject: [Freeswitch-users] Do Freeswitch conference support silent suppression? Message-ID: Currently I'm trying to create conference server using Freeswitch and hook up with the ROIP device (Vocality). The ROIP device is act as the voip extension for the Freeswitch, In order for the radio link to work, the conference must support the VAD/CNG so that the silence is not transmitted by the ROIP device to its radios.My problem is the ROIP will always transmit something on the radio. Am I correct to conclude that the Freeswitch not supporting silent suppression while in the conference mode.? thank you -nordin mohamed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/2bc97863/attachment.html From krice at freeswitch.org Wed May 8 02:52:40 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 07 May 2013 17:52:40 -0500 Subject: [Freeswitch-users] To the New People... Message-ID: Hey Guys, If you are new to the list, keep in mind that everyone joins the list moderated, and posting to the list the same email more then one time will just annoy the list moderators as they handle the moderated posts. Why do we moderate new users? Simple answer spam... Do you guys really want us to help the spammers? -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130507/f4f0e783/attachment.html From sertys at gmail.com Wed May 8 05:52:56 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Wed, 8 May 2013 03:52:56 +0200 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: <51897574.9080705@freeswitch.org> References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> <51897574.9080705@freeswitch.org> Message-ID: The guy is just asking a general question. And the general answer is - there are a ton of ways to achieve this. Natively via lua scripting, externally via ESL sockets and a script language of choice and dynamically with xml _curl generated dialplan instruction. FS is very, very controllable. On May 8, 2013 12:47 AM, "Raymond Chandler" wrote: > On 13-05-07 03:28 PM, Michael Collins wrote: > > And if you want it to be extremely dynamic you can send the originated > > leg into an xml_curl-generated dialplan or use mod_httapi to create > > something on the fly. > I think that's covered in the new book that's coming out soon. That > might be a worthwhile investment to make ;-) > > > -Ray > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130508/1748b93a/attachment.html From ibk at labhijau.net Wed May 8 06:08:34 2013 From: ibk at labhijau.net (Iwan Budi Kusnanto) Date: Wed, 8 May 2013 09:08:34 +0700 Subject: [Freeswitch-users] How to modify the way freeswitch handle SIP OPTIONS message In-Reply-To: References: Message-ID: On Wed, May 8, 2013 at 1:17 AM, Steven Ayre wrote: > That's the correct behaviour... what behaviour are you after? > > There's also a sofia profile parameter > sip-options-respond-503-on-busy. At the moment that's the only other > option. Hi Steve, Thanks. This is exactly what i'm looking for. >From the docs http://wiki.freeswitch.org/wiki/Sofia.conf.xml#sip-options-respond-503-on-busy, it will respond with SIP 503 when it reaches max-session limit. Can we set FS to send SIP 503 *before* it reaches max-session, say max-session - 100. > > -Steve > > > > On 7 May 2013 14:34, Iwan Budi Kusnanto wrote: >> Hi, >> I can see from console log that freeswitch always reply with SIP 200 >> when receive SIP OPTIONS message. >> Can i modify this behaviour? How to modify it? >> >> Thanks >> >> -- >> Iwan Budi Kusnanto >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Iwan Budi Kusnanto From spencer at 5ninesolutions.com Wed May 8 06:47:49 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 7 May 2013 19:47:49 -0700 Subject: [Freeswitch-users] Codec Negotiation Help In-Reply-To: <6f28692a07874746b81063da30587524@BLUPR04MB024.namprd04.prod.outlook.com> References: <69830897-FE1F-4A1F-AC0E-F5CC9DCE14A4@5ninesolutions.com> <6f28692a07874746b81063da30587524@BLUPR04MB024.namprd04.prod.outlook.com> Message-ID: <0F0ADCF5-4EE6-4E8E-9676-1F28B472B32C@5ninesolutions.com> Hi Robert, That was exactly what I was looking for. Even after I got the codecs offered the way I wanted, I found many of my endpoints didn't honor the preferences in the answer so even though I was offering PCMU,G729,G722,GSM and the phone was configured for G722,PCMU,G729 Cisco SPAs would answer with G722. So much for RFC3264... Thanks for the help! Spencer On May 7, 2013, at 1:37 PM, Robert Hadley wrote: > Hi Spencer, > > I also use late_negotiation and inherit_codec=true. I configure outbound codecs to append ep_codec_string using originator_codec. > > > > > > > > Regards, > Robert > > > -----Original Message----- > From: Spencer Thomason [mailto:spencer at 5ninesolutions.com] > Sent: Tuesday, May 07, 2013 9:03 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Codec Negotiation Help > > Hello all, > I'm having a bit of trouble getting codec negotiation working the way I'd like to. I'm trying to avoid transcoding unless its absolutely necessary for the call to be setup. I use two profiles with late negotiation enabled, an internal facing registered user endpoints and an external facing the PSTN via various ITSPs. For calls to users I've been setting codec_string=${ep_codec_string} in my dial strings and normal bridges work fine but I'm now reworking the hunt groups using enterprise originate and that doesn't seem to work correctly (the callee's preferences are always used). So I'd like to do the the "right" way now :-) > > The basic scenario is this: > > On the internal profile I have: > CODECS IN G722,PCMU,G729,GSM > CODECS OUT G722,PCMU,G729,GSM > > On the external profile I have: > CODECS IN G722,PCMU,G729 > CODECS OUT G722,PCMU,G729 > > What I'm trying to accomplish is this: > Call from PSTN, use caller's codecs as priority for the bridge, i.e. > PSTN GW --- PCMU, G729 --> FreeSWITCH --- PCMU,G729,G722, GSM ---> Local Endpoint PSTN GW --- G729,PCMU --> FreeSWITCH --- G729,PCMU,G722, GSM ---> Local Endpoint PSTN GW --- G722,PCMU --> FreeSWITCH --- G722,PCMU,G729, GSM ---> Local Endpoint > > > Call from local endpoint to PSTN, use callee's codecs as priority for the bridge, i.e. > Local Endpoint Offers G722, PCMU, G729 > GW Responds PCMU,G729 > Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU, G729 --> PSTN Desired negotiation: Both legs use PCMU > > Local Endpoint Offers G722, PCMU, G729 > GW Responds G729,PCMU > Local Endpoint --- G722,PCMU,G729 ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Both legs use G729 > > Local Endpoint Offers GSM > GW Responds PCMU, G729 > Local Endpoint --- GSM ---> FreeSWITCH --- G722,PCMU,G729 --> PSTN Desired negotiation: Leg A uses GSM, Leg B uses PCMU, FreeSWITCH transcodes. > > Any help is greatly appreciated! > > > Thanks, > Spencer > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From darcy at Vex.Net Wed May 8 07:33:52 2013 From: darcy at Vex.Net (D'Arcy J.M. Cain) Date: Tue, 7 May 2013 23:33:52 -0400 Subject: [Freeswitch-users] To the New People... In-Reply-To: References: Message-ID: <20130507233352.238bb97b@imp> On Tue, 07 May 2013 17:52:40 -0500 Ken Rice wrote: > If you are new to the list, keep in mind that everyone joins the list > moderated, and posting to the list the same email more then one time > will just annoy the list moderators as they handle the moderated > posts. I run all of my own lists the same way. It's an excellent idea. > Why do we moderate new users? Simple answer spam... Do you guys > really want us to help the spammers? No and I for one appreciate that this is one of the lists that I do not need to run through my spam filters. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net Voip: sip:darcy at Vex.Net From max at nysolutions.com Wed May 8 08:19:28 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 8 May 2013 04:19:28 +0000 Subject: [Freeswitch-users] Windows binary MSI installer Message-ID: The weekly generated MSI files are great and up to date, and include most 8khz sounds. Is there any particular reason not all 8khz audio files are included? Is it do to the file size? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/f2c38d77/attachment-0001.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/20130508/f2c38d77/attachment-0001.jpg From jaybinks at gmail.com Wed May 8 09:49:16 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 8 May 2013 15:49:16 +1000 Subject: [Freeswitch-users] Contact header, when using sofia gateways Message-ID: Im hitting an inter-op issue with a specific carrier ( running a Nortel CS2000 ) When I Send a call to this carrier using "Scenario A" everything works fine.. If I send them a call with "Scenario B" the call is rejected with a "502 bad gateway". the carrier has said that the contact header in "Scenario B" is "Too Long". Im not sure its the length specifically... but I wonder if the transport or gw variables are causing their switch to do something odd.. problem is, I cant seem to strip these with any combination of contact variables from the wiki or even after trawling /mod/endpoints/mod_sofia/* . it seems that gateway->register_contact is used for invites also. and the flow of parse_gateways in sofia.c suggests that there is no code flow that would avoid appending the gw= data. can someone confirm if there is a way avoid adding this extra data on calls out a gateway, or if im going to have to raise a request with the dev guys. *Scenario A )* I make an outbound call with this syntax : I See a SIP Contact of : *Scenario B )* I try and use gateways ( for the added functionality of monitoring the far ends reachability ) with this bridge string : I See a SIP Contact of : -- Sincerely Jay Binks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/cac088db/attachment.html From GB at cm.nl Wed May 8 11:09:09 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Wed, 8 May 2013 09:09:09 +0200 Subject: [Freeswitch-users] Generate dynamic dialplan based on variables In-Reply-To: References: <718850D1-113E-4CC2-B578-98733EEA579D@jerris.com> <1734A6D0-A2B8-46EA-941F-122320208A3F@jerris.com> <51897574.9080705@freeswitch.org> Message-ID: I'll take a deep dive into those methods. FreeSwitch is indeed very controllable and awesome! Thank guys! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel Ivanov Sent: Wednesday, May 8, 2013 3:53 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Generate dynamic dialplan based on variables The guy is just asking a general question. And the general answer is - there are a ton of ways to achieve this. Natively via lua scripting, externally via ESL sockets and a script language of choice and dynamically with xml _curl generated dialplan instruction. FS is very, very controllable. On May 8, 2013 12:47 AM, "Raymond Chandler" > wrote: On 13-05-07 03:28 PM, Michael Collins wrote: > And if you want it to be extremely dynamic you can send the originated > leg into an xml_curl-generated dialplan or use mod_httapi to create > something on the fly. I think that's covered in the new book that's coming out soon. That might be a worthwhile investment to make ;-) -Ray _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130508/9e58e0ca/attachment.html From mbodbg at gmx.net Wed May 8 12:55:31 2013 From: mbodbg at gmx.net (mbo) Date: Wed, 8 May 2013 10:55:31 +0200 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: References: Message-ID: <026C79B2-E1F2-4A71-AD71-A0BDCE409097@gmx.net> ok I see, I'll and update documentation in the wiki. I posted an other solution we found. Can you comment it? ... We did some more tests an found out that if we add a Event-UUID to the sendmsg command like: sendmsg Event-UUID: 5bf340cd-7a7e-4965-9285-95ed365ed242 call-command: execute execute-app-name: speak execute-app-arg: flite|slt|One We will get back this Event-UUID as Application-UUID in the CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE events, so we are able to match it. In our tests this works fine, however I'm not sure if this approach is reliable and has any side effects. May I ask for your opinion of this approach? ... Thanks Markus Am 07.05.2013 um 17:35 schrieb Anthony Minessale : > This is the first time the event is used. I changed to code in real time for you based on your request so you will be the one doing the documentation. That's how open source works. > > The point once you update and compile master, your actual sendmsg event that you send will now be sent back to you once it is processed with every header you supplied as well as the caller profile data from the channel you sent it on. This was what you asked for. > > > > On Sun, May 5, 2013 at 1:58 PM, mbo wrote: > Thanks for the answers. There is hardly any documentation of the PRIVATE_COMMAND event, can you give me some more details on it. What is it good for, when is it send, etc? > > If it is just an additional event including the channel data, where is the benefit using this event instead of set a custom variable first and then check this variable in the ChannelExecuteComplete event? Or am I missing something and it is possible to set a custom channel variable during SendMsg? > > Thanks > > Markus > > > Am 03.05.2013 um 21:13 schrieb Anthony Minessale : > >> Compromise: >> >> Update to latest and sub to the "private_command" event, you should get back you own event with channel data merged in. >> >> >> >> On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: >> >> >> On Fri, May 3, 2013 at 2:19 AM, mbo wrote: >> I'm referring to a two years old bug report http://jira.freeswitch.org/browse/FS-1309. >> >> Is it in the meantime somehow possible to map reply to SendMsg in asyc mode? I'm wondering why this ticket has been closed as "Won't fix", in my opinion is an essential feature to handle events properly. >> >> If not, I want to implement the work around described in the ticket, to set a channel variable in a round trip before executing the real command. Do I need to wait for the ChannelExecuteComplete event for the Set command, or can I send my "real" command right away after the Set command? >> To be safe you should verify that the set app actually ran before you implement the workaround. >> -MC >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130508/84ddc7a7/attachment-0001.html From yudha2008 at gmail.com Wed May 8 14:28:14 2013 From: yudha2008 at gmail.com (baskar) Date: Wed, 8 May 2013 03:28:14 -0700 (PDT) Subject: [Freeswitch-users] NO_ROUTE_DESTINATION Message-ID: <1368008894683-7590466.post@n2.nabble.com> Hi All, I have been testing freeswitch for more than a year. But now i am facing problem in new configuartion of SIP trunk. Here i have pasted by log http://pastebin.freeswitch.org/20882. When i try to call through gateway I execute through external profile. EXECUTE sofia/internal/407 at 10.15.1.41 bridge(sofia/gateway/xxxx/0018444485452) 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f] My dialplan default.xml Could some one help where i am wrong. Thanks in advance. Regards, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at askcornerstone.net Wed May 8 13:00:20 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Wed, 8 May 2013 09:00:20 +0000 Subject: [Freeswitch-users] External Softphone vs. Internal Question In-Reply-To: References: <80DFCBDE2AC6574487E3826FAF38F9CC387A7BC3@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387AAC51@vega.terisol.com> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CAB73@mail2.firstdataworks.net> As a follow up to this, I have another question! I got busy with some other things the last couple weeks and was revisiting this again. I realized that even though I thought I was registering to my external5090 profile, I was actually still registering to the internal one on 5060 (5060 is forwarded through my firewall to FS in addition to 5090). I discovered that this is because even though I put what I thought was the registration port 5090 in Jitsi (or Bria, or Linphone, etc.), the softphone was still registering to 5060. Couldn't figure out why, but then I put in the :5090 in the outbound proxy settings, and it worked and confirmed registered to 5090 in fs_cli. Why do I have to put the external ip in the proxy settings on my phone? Something set up wrong in my Freeswitch, my softphone(s)? Confused... Thanks, guys. Jeff From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, April 24, 2013 6:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] External Softphone vs. Internal Question The best way to learn more about this is Tony's "117" post: http://www.freeswitch.org/node/117 Also, check out Tony's "History of FreeSWITCH" in the FS book. (Note: we are nearly done with the 2e of the book, so don't buy the old one unless Packt gives you assurances that you can get the new one as well.) -MC On Wed, Apr 24, 2013 at 1:13 AM, Jeff Bernhardt > wrote: Thanks for taking the time to answer. I know it gets busy around here with all sorts of stuff that frankly is over my head! It's kind of nice that way, though... keeps some of the mystery and excitement alive for what's possible. Yeah, I didn't mean it like "Asterisk can do this so what the hell is wrong with Freeswitch?" Was just wondering why, so thanks for the clear explanation. I actually didn't know Asterisk had so much goofiness. Can you (or anyone else) give any examples of its goofiness? We're relatively light PBX users in general (just the basics for clients with no more than 150 phones, some with only 5 phones!), so we might not have come across any of them. Jeff Bernhardt Systems Administrator Cornerstone Consulting 808.440.2900 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, April 23, 2013 7:46 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] External Softphone vs. Internal Question Hi Jeff, The short answer is that you are not forced to create a separate profile for internal vs. external phones. However, FreeSWITCH gives you this freedom whereas Asterisk does not. You *could* try to cram everything into port 5060, but there's no compelling reason to do so. A lot of VoIPers are accustomed to using 5060 and only 5060, come what may. FreeSWITCHers generally view that as a limitation, not a feature. By having multiple SIP profiles - quite literally multiple SIP UAs - you have more freedom and flexibility to handle goofy scenarios like dealing with broken NAT devices. You can put all your broken stuff on a different profile and not have to worry that setting a particular option to fix one device will break another device. Oh, and keep in mind that "just because Asterisk can do it" doesn't mean that Asterisk does it correctly. There are a lot of devices out there that "work" but only because they all choose to be synchronized in their goofiness. Reams have been written about how FS does not pander to broken devices so I won't belabor the point here. Just know this: FS is relatively strict in adhering to specs and standards, so if something works with Asterisk (or whatever VoIP software) but not with FS then most likely it's a matter of figuring out how to tell FS to emulate the brokenness for the sake of interoperability. Hope this helps. Let us know how your setup is coming along. Be sure to use pastebin.freeswitch.org to share any configurations or logs with us. Thanks, -MC On Sat, Apr 20, 2013 at 2:50 AM, Jeff Bernhardt > wrote: Hi. I have the following basic setup questions: When using a softphone (Bria on iPhone) from external (on a different external ip address), I could register but no audio would be passed either way for any calls. I saw that I should set ext-rtp-ip in the internal sip profile to my external ip address (it was on auto-nat, which apparently wasn't working) in this wiki http://wiki.freeswitch.org/wiki/NAT_Traversal That didn't work, so I also set my ext-sip-ip to my public ip. After that, I could pass audio. However, if I register the phone internally instead and call for instance the IVR test line, the call drops after 30 seconds. So it's either no audio when registered externally or 30 second calls when registered internally. I found this wiki: http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios I fall into either scenario 2 or 3, and for both, it says to create a dedicated profile for external registrations and put them on port 5090, which works. However, is there no other way to solve this problem that doesn't require the use of an additional profile on port 5090 but also doesn't cut off internally registered calls after 30 seconds? On Asterisk, there's no need to open a second port to register external phones. What's different about Freeswitch? Also, I don't know what role these play, but I also get these errors: [WARNING] switch_core_media.c:1282 Asynchronous PTIME not supported, changing our end from 0 to 20 at seemingly random times ...and.... [INFO] switch_nat.c:590 NAT port mapping disabled when I make a call from internally or externally registered softphone to external number. Thank you. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/4418d8b6/attachment-0001.html From mehroz.ashraf85 at gmail.com Wed May 8 15:43:16 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Wed, 8 May 2013 04:43:16 -0700 (PDT) Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <1367907318527-7590411.post@n2.nabble.com> <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> Message-ID: <1368013396577-7590468.post@n2.nabble.com> Thanks Minessale , Setting up param name="rtp-timeout-sec" value="5" , in vars.xml , solved the issue. its working flawlessly! Thanks for all of yours contributions :) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590468.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ira at connectmevoice.com Wed May 8 16:06:38 2013 From: ira at connectmevoice.com (Ira Tessler) Date: Wed, 8 May 2013 08:06:38 -0400 Subject: [Freeswitch-users] DTMF Question/Issue Message-ID: I am having customer complaints that when they dial an outside conference bridge systems, (e.g. Go To Meeting, etc) or outside IVRs, they are having trouble with the remote system recognizing DTMF tones correctly (e.g. when entering a PIN for a conference bridge). I put in a way in our system to dial an outside number and bypass media for just that call. This seems to resolve the issue, but it is annoying for our customers. My question is how can I troubleshoot this issue? (I have looked at pcap's and from what I can see FS is setting the DTMF correctly) What DTMF settings in FS should I look at? Here is why I currently am using on my external sip profile: Thanks, Ira Tessler Lead Software Engineer ConnectMe (732) 490-9007 x2 ira at connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/c42dc016/attachment.html From jeff at jefflenk.com Wed May 8 17:50:17 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 8 May 2013 06:50:17 -0700 (PDT) Subject: [Freeswitch-users] Windows binary MSI installer In-Reply-To: References: Message-ID: <1368021017671-7590470.post@n2.nabble.com> What is missing? Is it a version problem? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590470.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveayre at gmail.com Wed May 8 18:15:03 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 8 May 2013 15:15:03 +0100 Subject: [Freeswitch-users] NO_ROUTE_DESTINATION In-Reply-To: <1368008894683-7590466.post@n2.nabble.com> References: <1368008894683-7590466.post@n2.nabble.com> Message-ID: A call to a gateway will leave from the Sofia profile that the gateway is configured on. -Steve On 8 May 2013 11:28, baskar wrote: > Hi All, > > I have been testing freeswitch for more than a year. But now i am facing > problem in new configuartion of SIP trunk. > > Here i have pasted by log http://pastebin.freeswitch.org/20882. > > When i try to call through gateway I execute through external profile. > > EXECUTE sofia/internal/407 at 10.15.1.41 > bridge(sofia/gateway/xxxx/0018444485452) > 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel > sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f] > > My dialplan default.xml > > > > data="continue_on_fail=CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER,NORMAL_UNSPECIFIED"/> > > > > > > Could some one help where i am wrong. > > Thanks in advance. > > Regards, > N.Baskar > > > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466.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://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 From max at nysolutions.com Wed May 8 18:43:19 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 8 May 2013 14:43:19 +0000 Subject: [Freeswitch-users] Windows binary MSI installer Message-ID: The one I noticed I am missing was voicemail/vm-enter_new_pin.wav, however after comparing to the recording download I saw the download has 91 files in that folder and the MSI only has 81 files. I can see based on the callie folder size that there are many more missing, not sure if those are needed. 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 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, May 08, 2013 9:50 AM To: freeswitch-users at lists.freeswitch.org Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer What is missing? Is it a version problem? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590470.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://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 From jleung at v10networks.ca Wed May 8 19:08:55 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 8 May 2013 08:08:55 -0700 Subject: [Freeswitch-users] Windows binary MSI installer In-Reply-To: References: Message-ID: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> More than likely those files are meant for voicemail. If you do need to use voicemail then you'll need that file so the prompt actually plays. The archive that has all of the prompts should be located here: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.15.tar.g z > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein > Sent: Wednesday, May 8, 2013 7:43 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Windows binary MSI installer > > The one I noticed I am missing was voicemail/vm-enter_new_pin.wav, > however after comparing to the recording download I saw the download has > 91 files in that folder and the MSI only has 81 files. I can see based on the > callie folder size that there are many more missing, not sure if those are > needed. > > > 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 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk > Sent: Wednesday, May 08, 2013 9:50 AM > To: freeswitch-users at lists.freeswitch.org > Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer > > What is missing? Is it a version problem? > > > > -- > View this message in context: http://freeswitch- > users.2379917.n2.nabble.com/Windows-binary-MSI-installer- > tp7590459p7590470.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://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 > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From asifk16 at gmail.com Wed May 8 13:00:06 2013 From: asifk16 at gmail.com (AKKy) Date: Wed, 8 May 2013 10:00:06 +0100 Subject: [Freeswitch-users] Problem with Spider Monkey In-Reply-To: References: Message-ID: Hi All I have a problem with connecting to a database from LUA using ODBC. I have noticed that when my freeswitch starts it doesn't load the spider monkey module. I did the following: Installed Freeswitch using debian repository from "deb http://files.freeswitch.org/repo/deb/debian/ squeeze main". I installed the package freeswitch-meta-vanilla and freeswitch-spidermonkey I copied the vanilla configuration from /usr/share/freeswitch/conf/vanilla/* to /etc/freeswitch folder I configured MySQL DB ODBC as described in http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core and tested it through isql command and it connected fine I started my freeswitch as root using the command /usr/bin/freeswitch and I got the below error 2013-05-07 22:19:33.922678 [CRIT] switch_loadable_module.c:1330 Error Loading module /usr/lib/freeswitch/mod/mod_spidermonkey.so **/usr/lib/freeswitch/mod/mod_spidermonkey.so: cannot open shared object file: No such file or directory** I didn't see the file mod_spidermonkey.so in my /usr/lib/freeswitch folder but in /opt/freeswitch/mod folder. So I added a symbolic link from /opt/freeswitch/mod to /usr/lib/freeswitch/mod for below files: mod_spidermonkey_core_db.so mod_spidermonkey_odbc.so mod_spidermonkey_socket.so mod_spidermonkey_curl.so mod_spidermonkey.so mod_spidermonkey_teletone.so Then I started freeswitch again and received the error below: 2013-05-07 22:30:25.813120 [CRIT] switch_loadable_module.c:1330 Error Loading module /usr/lib/freeswitch/mod/mod_spidermonkey.so **/usr/lib/freeswitch/mod/mod_spidermonkey.so: undefined symbol: switch_core_session_locate** Also when I try to execute a LUA script with the code freeswitch.Dbh("odbc://freeswitch") in a LUA file then I got the error 2013-05-07 23:05:46.915209 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not found, and no default driver specified 2013-05-07 23:05:46.915209 [CRIT] switch_core_sqldb.c:504 Failure to connect to PGSQL odbc://freeswitch! Any ideas what I am doing wrong? FYI - I am running on UBuntu Help will be appreciated. Regards Asif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/db95a8b6/attachment-0001.html From letterstack at gmail.com Wed May 8 13:30:28 2013 From: letterstack at gmail.com (Shiju V.Joseph) Date: Wed, 8 May 2013 15:00:28 +0530 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED Message-ID: Hi All, I have been experimenting with an odbc based freeswitch cluster in amazon ec2 , with opensips doing the load balancing function. I can make calls to mobile and landlines with out any issues with good quality voice , but when i try to call extension to extension freeswith shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot in lists and wiki and fs jira , tried out different dial strings but with out any results.Tried this one "dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & this ${sofia_contact(*/${dialed_user}@${dialed_domain})} I have copied the siptrace at http://pastebin.com/Qv79tjXK Appreciate any help in this regard Thanks -- shijujoe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/b3ec992b/attachment.html From rajat.toshniwal at tekmindz.com Wed May 8 12:36:33 2013 From: rajat.toshniwal at tekmindz.com (rajat) Date: Wed, 8 May 2013 01:36:33 -0700 (PDT) Subject: [Freeswitch-users] how Curl post dynamic variables Message-ID: <1368002192994-7590463.post@n2.nabble.com> Hi I am new to freeswitch and i am trying to integrate one php application with freeswitch. Purpose is freeswitch will send the user dialed (dtmf numbers) as input to this php application and in return it will send the response back. Dialplan which I have written for this is Now http://192.168.10.27/index.php this is a url on which i am sending my requests. Problem here is curl is not taking variable in the post. If instead of ${new}, i put some random static value like 256, then it is working perfectly fine but I want my users to input the values. Kindly suugest some way via which I can post variables in curl url -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-Curl-post-dynamic-variables-tp7590463.html Sent from the freeswitch-users mailing list archive at Nabble.com. From rajat.toshniwal at tekmindz.com Wed May 8 12:43:30 2013 From: rajat.toshniwal at tekmindz.com (Rajat toshniwal) Date: Wed, 08 May 2013 14:13:30 +0530 Subject: [Freeswitch-users] How to post dynamic variables Message-ID: <518A1032.3080904@tekmindz.com> Hi I am new to freeswitch and i am trying to integrate one php application with freeswitch. Purpose is freeswitch will send the user dialed (dtmf numbers) as input to this php application and in return it will send the response back. Dialplan which I have written for this is Now http://192.168.10.27/index.php this is a url on which i am sending my requests. Problem here is curl is not taking variable in the post. If instead of ${new}, i put some random static value like 256, then it is working perfectly fine but I want my users to input the values. Kindly suugest some way via which I can post variables in curl url ------------------------------------------------------------------------ ---------------------------------------------------------------------------------- Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd. ---------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/5bda9a1d/attachment.html From rajat.toshniwal at tekmindz.com Wed May 8 13:11:08 2013 From: rajat.toshniwal at tekmindz.com (rajat) Date: Wed, 8 May 2013 02:11:08 -0700 (PDT) Subject: [Freeswitch-users] curl issue with dynamic variables Message-ID: <1368004268786-7590465.post@n2.nabble.com> Hi I am new to freeswitch and i am trying to integrate one php application with freeswitch. Purpose is freeswitch will send the user dialed (dtmf numbers) as input to this php application and in return it will send the response back. Dialplan which I have written for this is Now http://192.168.10.27/index.php this is a url on which i am sending my requests. Problem here is curl is not taking variable in the post. If instead of ${new}, i put some random static value like 256, then it is working perfectly fine but I want my users to input the values. Kindly suugest some way via which I can post variables in curl url -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/curl-issue-with-dynamic-variables-tp7590465.html Sent from the freeswitch-users mailing list archive at Nabble.com. From max at nysolutions.com Wed May 8 19:22:36 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 8 May 2013 15:22:36 +0000 Subject: [Freeswitch-users] spam>spam>Re: Windows binary MSI installer In-Reply-To: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> References: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> Message-ID: Yes that fixed it, just wondering why you include 81 out of the 91 voicemail files, is it do to the file size, Or most user just don't need them? 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 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Leung Sent: Wednesday, May 08, 2013 11:09 AM To: 'FreeSWITCH Users Help' Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer More than likely those files are meant for voicemail. If you do need to use voicemail then you'll need that file so the prompt actually plays. The archive that has all of the prompts should be located here: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.15.tar.g z > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > Moishe Grunstein > Sent: Wednesday, May 8, 2013 7:43 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Windows binary MSI installer > > The one I noticed I am missing was voicemail/vm-enter_new_pin.wav, > however after comparing to the recording download I saw the download > has > 91 files in that folder and the MSI only has 81 files. I can see based > on the > callie folder size that there are many more missing, not sure if those > are needed. > > > 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 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > Jeff Lenk > Sent: Wednesday, May 08, 2013 9:50 AM > To: freeswitch-users at lists.freeswitch.org > Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer > > What is missing? Is it a version problem? > > > > -- > View this message in context: http://freeswitch- > users.2379917.n2.nabble.com/Windows-binary-MSI-installer- > tp7590459p7590470.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://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-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://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-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://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 From msc at freeswitch.org Wed May 8 19:26:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 08:26:30 -0700 Subject: [Freeswitch-users] DTMF Question/Issue In-Reply-To: References: Message-ID: Generally speaking it is useful to know what the far end is seeing. The DTMFs may leave your FS box in perfect health but they could be messed with along the way. Still, you can try setting your DTMF sending to in-band vs. RFC2833 and see if that helps. If not you'll probably need to talk to your provider and ask them what they see when the DTMFs pass through their network. -MC On Wed, May 8, 2013 at 5:06 AM, Ira Tessler wrote: > I am having customer complaints that when they dial an outside conference > bridge systems, (e.g. Go To Meeting, etc) or outside IVRs, they are having > trouble with the remote system recognizing DTMF tones correctly (e.g. when > entering a PIN for a conference bridge). I put in a way in our system to > dial an outside number and bypass media for just that call. This seems to > resolve the issue, but it is annoying for our customers. > > My question is how can I troubleshoot this issue? (I have looked at pcap's > and from what I can see FS is setting the DTMF correctly) What DTMF > settings in FS should I look at? Here is why I currently am using on my > external sip profile: > > > > > > > > > Thanks, > > Ira Tessler > Lead Software Engineer > ConnectMe > (732) 490-9007 x2 > ira at connectmevoice.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/3d0a5368/attachment-0001.html From msc at freeswitch.org Wed May 8 19:27:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 08:27:39 -0700 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED In-Reply-To: References: Message-ID: Go to fs_cli and manually run the sofia_contact command for your user. Does it find the user? -MC On Wed, May 8, 2013 at 2:30 AM, Shiju V.Joseph wrote: > Hi All, > > I have been experimenting with an odbc based freeswitch cluster in amazon > ec2 , with opensips doing the load balancing function. > > I can make calls to mobile and landlines with out any issues with good > quality voice , but when i try to call extension to extension freeswith > shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot > in lists and wiki and fs jira , tried out different dial strings but with > out any results.Tried this one > "dial-string" value="{presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & > this ${sofia_contact(*/${dialed_user}@${dialed_domain})} > > I have copied the siptrace at http://pastebin.com/Qv79tjXK > > Appreciate any help in this regard > > Thanks > -- > shijujoe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/597d06e1/attachment.html From krice at freeswitch.org Wed May 8 19:29:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 08 May 2013 10:29:05 -0500 Subject: [Freeswitch-users] Problem with Spider Monkey In-Reply-To: Message-ID: Open a jira on this sounds like a bug in the package On 5/8/13 4:00 AM, "AKKy" wrote: > > Hi All > > I have a problem with connecting to a database from LUA using ODBC. I have > noticed that when my freeswitch starts it doesn't load the spider monkey > module. I did the following: > > Installed Freeswitch using debian repository from "deb > http://files.freeswitch.org/repo/deb/debian/ squeeze main". I installed the > package freeswitch-meta-vanilla and freeswitch-spidermonkey > I copied the vanilla configuration from /usr/share/freeswitch/conf/vanilla/* > to /etc/freeswitch folder > I configured MySQL DB ODBC as described in > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core and tested it through > isql command and it connected fine > I started my freeswitch as root using the command /usr/bin/freeswitch and I > got the below error > 2013-05-07 22:19:33.922678 [CRIT] switch_loadable_module.c:1330 Error Loading > module /usr/lib/freeswitch/mod/mod_spidermonkey.so > **/usr/lib/freeswitch/mod/mod_spidermonkey.so: cannot open shared object file: > No such file or directory** > > I didn't see the file mod_spidermonkey.so in my /usr/lib/freeswitch folder but > in /opt/freeswitch/mod folder. So I added a symbolic link from > /opt/freeswitch/mod to /usr/lib/freeswitch/mod for below files: > ????? mod_spidermonkey_core_db.so > ????? mod_spidermonkey_odbc.so? > ????? mod_spidermonkey_socket.so > ????? mod_spidermonkey_curl.so > ????? mod_spidermonkey.so > ????? mod_spidermonkey_teletone.so > > Then I started freeswitch again and received the error below: > 2013-05-07 22:30:25.813120 [CRIT] switch_loadable_module.c:1330 Error Loading > module /usr/lib/freeswitch/mod/mod_spidermonkey.so > **/usr/lib/freeswitch/mod/mod_spidermonkey.so: undefined symbol: > switch_core_session_locate** > > Also when I try to execute a LUA script with the code > freeswitch.Dbh("odbc://freeswitch") in a LUA file then I got the error > > 2013-05-07 23:05:46.915209 [ERR] switch_odbc.c:365 STATE: IM002 CODE 0 ERROR: > [unixODBC][Driver Manager]Data source name not found, and no default driver > specified > 2013-05-07 23:05:46.915209 [CRIT] switch_core_sqldb.c:504 Failure to connect > to PGSQL odbc://freeswitch! > > Any ideas what I am doing wrong? FYI - I am running on UBuntu > > Help will be appreciated. > > Regards > Asif > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/7116c57c/attachment.html From spencer at 5ninesolutions.com Wed May 8 19:30:59 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 8 May 2013 08:30:59 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP Message-ID: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> Hello all, In our deployments, Freeswitch is on a public IP with most user endpoints behind NAT. As the demand for BLFs has grown I've been forced to go to TCP due the NOTIFYs exceeding MTU. I've been reluctant to use TCP for SIP due to the increased overhead. Currently I force a registration expiry of 600 seconds and ping all NATed endpoints. I'm exploring the idea of switching all endpoints to TCP where available, ditching the options pings and dropping the registration expiration to 300 secs which should exceed the TCP connection timeout of almost every router I've seen. I was currious if anyone had an experience or could point out any caveats I might run into. Thanks in advance, Spencer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/7541b40a/attachment.html From jleung at v10networks.ca Wed May 8 19:34:29 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 8 May 2013 08:34:29 -0700 Subject: [Freeswitch-users] spam>spam>Re: Windows binary MSI installer In-Reply-To: References: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> Message-ID: <00bc01ce4c01$88d96570$9a8c3050$@v10networks.ca> More than likely the WiX project file wasn't updated to include the added sound files. > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Moishe Grunstein > Sent: Wednesday, May 8, 2013 8:23 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] spam>spam>Re: Windows binary MSI > installer > > Yes that fixed it, just wondering why you include 81 out of the 91 voicemail > files, is it do to the file size, Or most user just don't need them? > > > 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 > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Jeff Leung > Sent: Wednesday, May 08, 2013 11:09 AM > To: 'FreeSWITCH Users Help' > Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer > > More than likely those files are meant for voicemail. If you do need to use > voicemail then you'll need that file so the prompt actually plays. The archive > that has all of the prompts should be located here: > http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.15.tar.g > z > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > > Moishe Grunstein > > Sent: Wednesday, May 8, 2013 7:43 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Windows binary MSI installer > > > > The one I noticed I am missing was voicemail/vm-enter_new_pin.wav, > > however after comparing to the recording download I saw the download > > has > > 91 files in that folder and the MSI only has 81 files. I can see based > > on > the > > callie folder size that there are many more missing, not sure if those > > are needed. > > > > > > 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 > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > > Jeff Lenk > > Sent: Wednesday, May 08, 2013 9:50 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer > > > > What is missing? Is it a version problem? > > > > > > > > -- > > View this message in context: http://freeswitch- > > users.2379917.n2.nabble.com/Windows-binary-MSI-installer- > > tp7590459p7590470.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://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-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://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-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://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 > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From jleung at v10networks.ca Wed May 8 19:37:53 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 8 May 2013 08:37:53 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> Message-ID: <00bd01ce4c02$02281570$06784050$@v10networks.ca> On a Linux system there is a limit of how many open TCP connections you have. Unless you have a crazy amount of endpoints you have to serve, TCP probably isn?t really worth it in my opinion. Also did I also mention that TCP connections don?t really fix NAT issues? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Spencer Thomason Sent: Wednesday, May 8, 2013 8:31 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] TCP vs UDP SIP Hello all, In our deployments, Freeswitch is on a public IP with most user endpoints behind NAT. As the demand for BLFs has grown I've been forced to go to TCP due the NOTIFYs exceeding MTU. I've been reluctant to use TCP for SIP due to the increased overhead. Currently I force a registration expiry of 600 seconds and ping all NATed endpoints. I'm exploring the idea of switching all endpoints to TCP where available, ditching the options pings and dropping the registration expiration to 300 secs which should exceed the TCP connection timeout of almost every router I've seen. I was currious if anyone had an experience or could point out any caveats I might run into. Thanks in advance, Spencer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/ed5bbc7b/attachment.html From bedgar at vseinc.com Wed May 8 19:41:29 2013 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Wed, 8 May 2013 11:41:29 -0400 Subject: [Freeswitch-users] Artificial Rings on Inbound TDM calls Message-ID: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> Hello, Is there any way to present an artificial ring or two to the caller while receiving and processing the DNIS and ANI on an inbound TDM call? There is nearly a 5 second delay during that phase of the call. Thank you, Brian C. Edgar, Jr. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/c03b91a0/attachment.html From bpriddy at bryantschools.org Wed May 8 19:48:12 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 8 May 2013 10:48:12 -0500 Subject: [Freeswitch-users] Call Rejected Message-ID: I was having this problem on one phone. Now another one is doing it. I place a call out and it says denied on the phone. In the log is a call rejected.. Link to FS console Internal calls to and from are fine! But calling out is the issue. http://pastebin.com/SkLQ9sHE -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/d45a9793/attachment.html From jeff at jefflenk.com Wed May 8 19:49:40 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 8 May 2013 08:49:40 -0700 (PDT) Subject: [Freeswitch-users] Windows binary MSI installer In-Reply-To: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> References: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> Message-ID: <1368028180930-7590482.post@n2.nabble.com> Ok I reset the build for sound file inclusion so a new msi should be available this weekend. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590482.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Wed May 8 19:49:26 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 08 May 2013 10:49:26 -0500 Subject: [Freeswitch-users] FreeSWITCH Weekly Community Conference Call - Today CGRateS Message-ID: Hey Guys, don?t forget coming up at 1PM EST (10AM PST) is the Weekly FreeSWITCH Community Conference Call. This week we have Dan Christian Bogos joining us to talk about CGRateS, a FreeSWITCH Compatible rating and billing system. More info on the wiki link including bridge access information. http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_08 TL;DR version: Call SIP:888 at conference.freeswitch.org at 1PM Eastern -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/bfa81f7b/attachment.html From krice at freeswitch.org Wed May 8 19:52:27 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 08 May 2013 10:52:27 -0500 Subject: [Freeswitch-users] Call Rejected In-Reply-To: Message-ID: Enable Sip tracing but it looks like your upstream is 603ing the call On 5/8/13 10:48 AM, "Blake Priddy" wrote: > I was having this problem on one phone. Now another one is doing it. I place a > call out and it says denied on the phone. In the log is a call rejected.. > > Link to FS console > > Internal calls to and from are fine! But calling out is the issue. > > http://pastebin.com/SkLQ9sHE -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/ef2ce69d/attachment.html From max at nysolutions.com Wed May 8 20:00:28 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 8 May 2013 16:00:28 +0000 Subject: [Freeswitch-users] spam>spam>Re: Windows binary MSI installer In-Reply-To: <1368028180930-7590482.post@n2.nabble.com> References: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> <1368028180930-7590482.post@n2.nabble.com> Message-ID: Thanks, greatly appreciated. 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 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, May 08, 2013 11:50 AM To: freeswitch-users at lists.freeswitch.org Subject: spam>spam>Re: [Freeswitch-users] Windows binary MSI installer Ok I reset the build for sound file inclusion so a new msi should be available this weekend. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590482.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://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 From vipkilla at gmail.com Wed May 8 20:17:57 2013 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 8 May 2013 12:17:57 -0400 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <00bd01ce4c02$02281570$06784050$@v10networks.ca> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> Message-ID: In my opinion, TCP seems better than UDP as you know all the SIP packets are making to their destination. On Wed, May 8, 2013 at 11:37 AM, Jeff Leung wrote: > On a Linux system there is a limit of how many open TCP connections you > have. > I never heard this before...where and how it this limit defined? > Unless you have a crazy amount of endpoints you have to serve, TCP > probably isn?t really worth it in my opinion. > How many endpoints? > Also did I also mention that TCP connections don?t really fix NAT issues? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/136581b9/attachment.html From bpriddy at bryantschools.org Wed May 8 20:21:18 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 8 May 2013 11:21:18 -0500 Subject: [Freeswitch-users] Call Rejected In-Reply-To: References: Message-ID: Here it is with trace http://pastebin.com/Nbfmk7bU On Wed, May 8, 2013 at 10:52 AM, Ken Rice wrote: > Enable Sip tracing but it looks like your upstream is 603ing the call > > > > On 5/8/13 10:48 AM, "Blake Priddy" wrote: > > I was having this problem on one phone. Now another one is doing it. I > place a call out and it says denied on the phone. In the log is a call > rejected.. > > Link to FS console > > Internal calls to and from are fine! But calling out is the issue. > > http://pastebin.com/SkLQ9sHE > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/29081cd9/attachment.html From msc at freeswitch.org Wed May 8 20:32:35 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 09:32:35 -0700 Subject: [Freeswitch-users] Call Rejected In-Reply-To: References: Message-ID: Ken is correct, the carrier is sending back a 603: ------------------------------------------------------------------------ recv 475 bytes from udp/[216.246.105.146]:5060 at 16:19:51.798622: ------------------------------------------------------------------------ SIP/2.0 603 Declined Via: SIP/2.0/UDP 10.32.4.2:5080 ;branch=z9hG4bKepUQ0tKj9gmcj;received=99.43.25.66;rport=65496 From: " Mrs. Johnston" ;tag=51v2r6Q9NKNra To: ;tag=as328d2593 Call-ID: f3bcffa2-329d-1231-149a-0019b9ee5254 CSeq: 43678932 INVITE Server: Viatalk SIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 You'll need to contact the carrier and find out why they don't like this call. -MC On Wed, May 8, 2013 at 9:21 AM, Blake Priddy wrote: > Here it is with trace > > http://pastebin.com/Nbfmk7bU > > > On Wed, May 8, 2013 at 10:52 AM, Ken Rice wrote: > >> Enable Sip tracing but it looks like your upstream is 603ing the call >> >> >> >> On 5/8/13 10:48 AM, "Blake Priddy" wrote: >> >> I was having this problem on one phone. Now another one is doing it. I >> place a call out and it says denied on the phone. In the log is a call >> rejected.. >> >> Link to FS console >> >> Internal calls to and from are fine! But calling out is the issue. >> >> http://pastebin.com/SkLQ9sHE >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/6bdb40e4/attachment.html From mike at jerris.com Wed May 8 20:33:05 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 8 May 2013 12:33:05 -0400 Subject: [Freeswitch-users] Contact header, when using sofia gateways In-Reply-To: References: Message-ID: <7D3DAED9-458F-4F0C-8AE1-F5F3479BB88D@jerris.com> No way to turn it off right now. If you want some quick hacking to confirm, check out in stable sofia.c around line 3156 if (! zstr(contact_params)) { the code in that if block sets the contact params. would be trivial enough to add some options to tweak that behavior but probably best to figure out what the real issue is first. Mike On May 8, 2013, at 1:49 AM, jay binks wrote: > Im hitting an inter-op issue with a specific carrier ( running a Nortel CS2000 ) > > When I Send a call to this carrier using "Scenario A" everything works fine.. > > If I send them a call with "Scenario B" the call is rejected with a "502 bad gateway". > the carrier has said that the contact header in "Scenario B" is "Too Long". Im not sure its the length specifically... but I wonder if the transport or gw variables are causing their switch to do something odd.. > > problem is, I cant seem to strip these with any combination of contact variables from the wiki or even after trawling /mod/endpoints/mod_sofia/* . > > it seems that gateway->register_contact is used for invites also. > and the flow of parse_gateways in sofia.c suggests that there is no code flow that would avoid appending the gw= data. > > can someone confirm if there is a way avoid adding this extra data on calls out a gateway, > or if im going to have to raise a request with the dev guys. > > Scenario A ) > I make an outbound call with this syntax : > > > I See a SIP Contact of : > > > > Scenario B ) > I try and use gateways ( for the added functionality of monitoring the far ends reachability ) > with this bridge string : > > > I See a SIP Contact of : > > > -- > Sincerely > > Jay Binks > _________________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/948c03dd/attachment-0001.html From msc at freeswitch.org Wed May 8 20:34:59 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 09:34:59 -0700 Subject: [Freeswitch-users] How to post dynamic variables In-Reply-To: <518A1032.3080904@tekmindz.com> References: <518A1032.3080904@tekmindz.com> Message-ID: On Wed, May 8, 2013 at 1:43 AM, Rajat toshniwal < rajat.toshniwal at tekmindz.com> wrote: > ** > Hi > > I am new to freeswitch and i am trying to integrate one php application > with freeswitch. Purpose is freeswitch will send the user dialed (dtmf > numbers) as input to this php application and in return it will send the > response back. > > Dialplan which I have written for this is > > > > > > > > > > > > > > > > > > Now http://192.168.10.27/index.php this is a url on which i am sending my > requests. Problem here is curl is not taking variable in the post. If > instead of ${new}, i put some random static value like 256, then it is > working perfectly fine but I want my users to input the values. Kindly > suugest some way via which I can post variables in curl url > The issue is that you have inline="true" on your curl app. Remove that and you should be fine. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/13e1e810/attachment.html From krice at freeswitch.org Wed May 8 20:42:44 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 08 May 2013 11:42:44 -0500 Subject: [Freeswitch-users] Call Rejected In-Reply-To: Message-ID: See line 371 of the trace and contact them to complain and ask why they are rejecting it On 5/8/13 10:48 AM, "Blake Priddy" wrote: > I was having this problem on one phone. Now another one is doing it. I place a > call out and it says denied on the phone. In the log is a call rejected.. > > Link to FS console > > Internal calls to and from are fine! But calling out is the issue. > > http://pastebin.com/SkLQ9sHE -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/ab05b9b4/attachment.html From bpriddy at bryantschools.org Wed May 8 20:43:17 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 8 May 2013 11:43:17 -0500 Subject: [Freeswitch-users] Call Rejected In-Reply-To: References: Message-ID: Ken is always correct!!! ;) On Wed, May 8, 2013 at 11:32 AM, Michael Collins wrote: > Ken is correct, the carrier is sending back a 603: > > > ------------------------------------------------------------------------ > recv 475 bytes from udp/[216.246.105.146]:5060 at 16:19:51.798622: > ------------------------------------------------------------------------ > SIP/2.0 603 Declined > Via: SIP/2.0/UDP 10.32.4.2:5080 > ;branch=z9hG4bKepUQ0tKj9gmcj;received=99.43.25.66;rport=65496 > From: " Mrs. Johnston" >;tag=51v2r6Q9NKNra > To: ;tag=as328d2593 > Call-ID: f3bcffa2-329d-1231-149a-0019b9ee5254 > CSeq: 43678932 INVITE > Server: Viatalk SIP > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 > > You'll need to contact the carrier and find out why they don't like this > call. > -MC > > > > On Wed, May 8, 2013 at 9:21 AM, Blake Priddy wrote: > >> Here it is with trace >> >> http://pastebin.com/Nbfmk7bU >> >> >> On Wed, May 8, 2013 at 10:52 AM, Ken Rice wrote: >> >>> Enable Sip tracing but it looks like your upstream is 603ing the call >>> >>> >>> >>> On 5/8/13 10:48 AM, "Blake Priddy" wrote: >>> >>> I was having this problem on one phone. Now another one is doing it. I >>> place a call out and it says denied on the phone. In the log is a call >>> rejected.. >>> >>> Link to FS console >>> >>> Internal calls to and from are fine! But calling out is the issue. >>> >>> http://pastebin.com/SkLQ9sHE >>> >>> >>> -- >>> Ken >>> *http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> *irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> >> *Blakelund Priddy* >> Network & Systems Engineer >> Bryant Public School District >> Bryant, Arkansas 72022 >> http://www.bryantschools.org >> p 501-653-5038 >> f 501-847-5656 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/e88f73dc/attachment-0001.html From msc at freeswitch.org Wed May 8 20:44:21 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 09:44:21 -0700 Subject: [Freeswitch-users] External Softphone vs. Internal Question In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CAB73@mail2.firstdataworks.net> References: <80DFCBDE2AC6574487E3826FAF38F9CC387A7BC3@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387AAC51@vega.terisol.com> <8A9716A5B256904FB1F07C050F9CCCCB020CAB73@mail2.firstdataworks.net> Message-ID: On Wed, May 8, 2013 at 2:00 AM, Jeff Bernhardt wrote: > As a follow up to this, I have another question! I got busy with some > other things the last couple weeks and was revisiting this again. I > realized that even though I thought I was registering to my external5090 > profile, I was actually still registering to the internal one on 5060 (5060 > is forwarded through my firewall to FS in addition to 5090). I discovered > that this is because even though I put what I thought was the registration > port 5090 in Jitsi (or Bria, or Linphone, etc.), the softphone was still > registering to 5060. Couldn?t figure out why, but then I put in the > :5090 in the outbound proxy settings, and it worked and > confirmed registered to 5090 in fs_cli. Why do I have to put the external > ip in the proxy settings on my phone? Something set up wrong in my > Freeswitch, my softphone(s)? Confused? > Without knowing the topology we are equally confused. Is your phone behind the same NAT device as FreeSWITCH? What happens when you put :5090 in the proxy settings of the phone? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/c031bc1d/attachment.html From jleung at v10networks.ca Wed May 8 20:51:02 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 8 May 2013 09:51:02 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> Message-ID: <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Vik Killa > Sent: Wednesday, May 8, 2013 9:18 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] TCP vs UDP SIP > That I would agree with, but the thing is you lose the capability of failover in the unlikely event that a node in a FreeSWITCH cluster fail. > In my opinion, TCP seems better than UDP as you know all the SIP packets > are making to their destination. > > On Wed, May 8, 2013 at 11:37 AM, Jeff Leung > wrote: > > > On a Linux system there is a limit of how many open TCP connections > you have. > If I can remember correctly, I think Darren from 2600hz did discuss about the limit of open TCP connections you can have on a Linux system. Correct me if I'm wrong on this, but that seems to be the case. And I have seen instances of that happening on a misconfigured Squid Proxy > I never heard this before...where and how it this limit defined? > > > Unless you have a crazy amount of endpoints you have to serve, TCP > probably isn't really worth it in my opinion. > > Assuming it's one Open TCP connection per endpoint, you'd probably need more endpoints than the maximum amount of open TCP connections to hit that problem > How many endpoints? > > > Also did I also mention that TCP connections don't really fix NAT > issues? > > From jmesquita at freeswitch.org Wed May 8 21:09:22 2013 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 8 May 2013 14:09:22 -0300 Subject: [Freeswitch-users] Artificial Rings on Inbound TDM calls In-Reply-To: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> References: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> Message-ID: <23DEA328-FE31-4258-84BD-443F52EB6529@freeswitch.org> Ring_ready should do that right? Sent from my iPhone On May 8, 2013, at 12:41 PM, wrote: > Hello, > > Is there any way to present an artificial ring or two to the caller while receiving and processing the DNIS and ANI on an inbound TDM call? There is nearly a 5 second delay during that phase of the call. > > Thank you, > > Brian C. Edgar, Jr. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130508/c3462ee2/attachment.html From spencer at 5ninesolutions.com Wed May 8 21:13:10 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 8 May 2013 10:13:10 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> Message-ID: <2BF00B16-7505-4D5B-8E83-24AE07BA857D@5ninesolutions.com> Hi Jeff, Thanks for the insight. Forgive my ignorance but if I have two Identical Freeswitch servers with SRV records and endpoints that properly support SRVs, why do I loose the ability to failover if one host is not reachable? Also as many of these end points are Polycoms behind NAT, I can't see any reason I'd still need NDLB-force-rport on the profile? Since these are application servers, handling conferences, presence, etc., I'd imagine I would hit other bottlenecks before I hit the TCP connection limit. On May 8, 2013, at 9:51 AM, "Jeff Leung" wrote: > > >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- >> users-bounces at lists.freeswitch.org] On Behalf Of Vik Killa >> Sent: Wednesday, May 8, 2013 9:18 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] TCP vs UDP SIP > > That I would agree with, but the thing is you lose the capability of > failover in the unlikely event that a node in a FreeSWITCH cluster fail. > >> In my opinion, TCP seems better than UDP as you know all the SIP packets >> are making to their destination. >> >> On Wed, May 8, 2013 at 11:37 AM, Jeff Leung >> wrote: >> >> >> On a Linux system there is a limit of how many open TCP connections >> you have. > > If I can remember correctly, I think Darren from 2600hz did discuss about > the limit of open TCP connections you can have on a Linux system. Correct me > if I'm wrong on this, but that seems to be the case. And I have seen > instances of that happening on a misconfigured Squid Proxy > >> I never heard this before...where and how it this limit defined? >> >> >> Unless you have a crazy amount of endpoints you have to serve, TCP >> probably isn't really worth it in my opinion. > > Assuming it's one Open TCP connection per endpoint, you'd probably need > more endpoints than the maximum amount of open TCP connections to hit that > problem > >> How many endpoints? >> >> >> Also did I also mention that TCP connections don't really fix NAT >> issues? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From anthony.minessale at gmail.com Wed May 8 21:16:56 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 May 2013 12:16:56 -0500 Subject: [Freeswitch-users] Correlate SendMsg reply with request in async mode In-Reply-To: <026C79B2-E1F2-4A71-AD71-A0BDCE409097@gmx.net> References: <026C79B2-E1F2-4A71-AD71-A0BDCE409097@gmx.net> Message-ID: Yes this is fine as well. On Wed, May 8, 2013 at 3:55 AM, mbo wrote: > ok I see, I'll and update documentation in the wiki. I posted an other > solution we found. Can you comment it? > > ... > > We did some more tests an found out that if we add a Event-UUID to the > sendmsg command like: > > sendmsg > Event-UUID: 5bf340cd-7a7e-4965-9285-95ed365ed242 > call-command: execute > execute-app-name: speak > execute-app-arg: flite|slt|One > > We will get back this Event-UUID as Application-UUID in the > CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE events, so we are able to > match it. In our tests this works fine, however I'm not sure if this > approach is reliable and has any side effects. May I ask for your opinion > of this approach? > > ... > > Thanks > > Markus > > > > Am 07.05.2013 um 17:35 schrieb Anthony Minessale < > anthony.minessale at gmail.com>: > > This is the first time the event is used. I changed to code in real time > for you based on your request so you will be the one doing the > documentation. That's how open source works. > > The point once you update and compile master, your actual sendmsg event > that you send will now be sent back to you once it is processed with every > header you supplied as well as the caller profile data from the channel you > sent it on. This was what you asked for. > > > > On Sun, May 5, 2013 at 1:58 PM, mbo wrote: > >> Thanks for the answers. There is hardly any documentation of the >> PRIVATE_COMMAND event, can you give me some more details on it. What is it >> good for, when is it send, etc? >> >> If it is just an additional event including the channel data, where is >> the benefit using this event instead of set a custom variable first and >> then check this variable in the ChannelExecuteComplete event? Or am I >> missing something and it is possible to set a custom channel variable >> during SendMsg? >> >> Thanks >> >> Markus >> >> >> Am 03.05.2013 um 21:13 schrieb Anthony Minessale < >> anthony.minessale at gmail.com>: >> >> Compromise: >> >> Update to latest and sub to the "private_command" event, you should get >> back you own event with channel data merged in. >> >> >> >> On Fri, May 3, 2013 at 12:27 PM, Michael Collins wrote: >> >>> >>> >>> On Fri, May 3, 2013 at 2:19 AM, mbo wrote: >>> >>>> I'm referring to a two years old bug report >>>> http://jira.freeswitch.org/browse/FS-1309. >>>> >>>> Is it in the meantime somehow possible to map reply to SendMsg in asyc >>>> mode? I'm wondering why this ticket has been closed as "Won't fix", in my >>>> opinion is an essential feature to handle events properly. >>>> >>>> If not, I want to implement the work around described in the ticket, to >>>> set a channel variable in a round trip before executing the real command. >>>> Do I need to wait for the ChannelExecuteComplete event for the Set command, >>>> or can I send my "real" command right away after the Set command? >>>> >>> To be safe you should verify that the set app actually ran before you >>> implement the workaround. >>> -MC >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/4c208ea3/attachment-0001.html From jleung at v10networks.ca Wed May 8 21:20:43 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Wed, 8 May 2013 10:20:43 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <2BF00B16-7505-4D5B-8E83-24AE07BA857D@5ninesolutions.com> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> <2BF00B16-7505-4D5B-8E83-24AE07BA857D@5ninesolutions.com> Message-ID: <000601ce4c10$5fad6570$1f083050$@v10networks.ca> > Hi Jeff, > Thanks for the insight. Forgive my ignorance but if I have two Identical > Freeswitch servers with SRV records and endpoints that properly support > SRVs, why do I loose the ability to failover if one host is not reachable? TCP is a stateful protocol. On the other hand UDP isn't, it's stateless. It's just easier to failover with UDP than with TCP if you understand the difference between the two protocols. I'm not saying that it's not possible to do so with TCP, but with the way how SIP works, you'd want to use UDP if you want failover capabilities without the headache. > Also as many of these end points are Polycoms behind NAT, I can't see any > reason I'd still need NDLB-force-rport on the profile? > Unfortunately, I don't work with Polycom phones. Brian West over here can comment on that issue. > Since these are application servers, handling conferences, presence, etc., I'd > imagine I would hit other bottlenecks before I hit the TCP connection limit. Yes that's true, but if you had a FreeSWITCH box that purely handled SIP messages and no media, you'd probably hit that TCP Open connection limit. > On May 8, 2013, at 9:51 AM, "Jeff Leung" wrote: > > > > > > >> -----Original Message----- > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > >> Vik Killa > >> Sent: Wednesday, May 8, 2013 9:18 AM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] TCP vs UDP SIP > > > > That I would agree with, but the thing is you lose the capability of > > failover in the unlikely event that a node in a FreeSWITCH cluster fail. > > > >> In my opinion, TCP seems better than UDP as you know all the SIP > >> packets are making to their destination. > >> > >> On Wed, May 8, 2013 at 11:37 AM, Jeff Leung > >> wrote: > >> > >> > >> On a Linux system there is a limit of how many open TCP > >> connections you have. > > > > If I can remember correctly, I think Darren from 2600hz did discuss > > about the limit of open TCP connections you can have on a Linux > > system. Correct me if I'm wrong on this, but that seems to be the > > case. And I have seen instances of that happening on a misconfigured > > Squid Proxy > > > >> I never heard this before...where and how it this limit defined? > >> > >> > >> Unless you have a crazy amount of endpoints you have to serve, TCP > >> probably isn't really worth it in my opinion. > > > > Assuming it's one Open TCP connection per endpoint, you'd probably > > need more endpoints than the maximum amount of open TCP connections > to > > hit that problem > > > >> How many endpoints? > >> > >> > >> Also did I also mention that TCP connections don't really fix NAT > >> issues? > > > > > > > > > __________________________________________________________ > ____________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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-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://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 From spencer at 5ninesolutions.com Wed May 8 22:13:10 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 8 May 2013 11:13:10 -0700 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <000601ce4c10$5fad6570$1f083050$@v10networks.ca> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> <2BF00B16-7505-4D5B-8E83-24AE07BA857D@5ninesolutions.com> <000601ce4c10$5fad6570$1f083050$@v10networks.ca> Message-ID: <06BFB360-C378-46AE-B894-C7552966B241@5ninesolutions.com> Understood. My plan is to use UDP for all "trunking" type endpoints at TCP for desk phones as they will likely receive more NOTIFYs and in most cases being behind NAT where the longer connection timeout comes in handy. I also found a good paper on the subject: http://www.cs.columbia.edu/~kumiko/publish/IPTComm08_paper.pdf In regard to connection timeout how does Freeswitch handle this? I noticed the new Sofia parameters and I was curious if the connection lifetime was configurable as well. BR, Spencer On May 8, 2013, at 10:20 AM, "Jeff Leung" wrote: >> Hi Jeff, >> Thanks for the insight. Forgive my ignorance but if I have two Identical >> Freeswitch servers with SRV records and endpoints that properly support >> SRVs, why do I loose the ability to failover if one host is not reachable? > > TCP is a stateful protocol. On the other hand UDP isn't, it's stateless. > It's just easier to failover with UDP than with TCP if you understand the > difference between the two protocols. I'm not saying that it's not possible > to do so with TCP, but with the way how SIP works, you'd want to use UDP if > you want failover capabilities without the headache. > >> Also as many of these end points are Polycoms behind NAT, I can't see any >> reason I'd still need NDLB-force-rport on the profile? > > Unfortunately, I don't work with Polycom phones. Brian West over here can > comment on that issue. > >> Since these are application servers, handling conferences, presence, etc., > I'd >> imagine I would hit other bottlenecks before I hit the TCP connection > limit. > > Yes that's true, but if you had a FreeSWITCH box that purely handled SIP > messages and no media, you'd probably hit that TCP Open connection limit. > >> On May 8, 2013, at 9:51 AM, "Jeff Leung" wrote: >> >>> >>> >>>> -----Original Message----- >>>> From: freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of >>>> Vik Killa >>>> Sent: Wednesday, May 8, 2013 9:18 AM >>>> To: FreeSWITCH Users Help >>>> Subject: Re: [Freeswitch-users] TCP vs UDP SIP >>> >>> That I would agree with, but the thing is you lose the capability of >>> failover in the unlikely event that a node in a FreeSWITCH cluster fail. >>> >>>> In my opinion, TCP seems better than UDP as you know all the SIP >>>> packets are making to their destination. >>>> >>>> On Wed, May 8, 2013 at 11:37 AM, Jeff Leung >>>> wrote: >>>> >>>> >>>> On a Linux system there is a limit of how many open TCP >>>> connections you have. >>> >>> If I can remember correctly, I think Darren from 2600hz did discuss >>> about the limit of open TCP connections you can have on a Linux >>> system. Correct me if I'm wrong on this, but that seems to be the >>> case. And I have seen instances of that happening on a misconfigured >>> Squid Proxy >>> >>>> I never heard this before...where and how it this limit defined? >>>> >>>> >>>> Unless you have a crazy amount of endpoints you have to serve, TCP >>>> probably isn't really worth it in my opinion. >>> >>> Assuming it's one Open TCP connection per endpoint, you'd probably >>> need more endpoints than the maximum amount of open TCP connections >> to >>> hit that problem >>> >>>> How many endpoints? >>>> >>>> >>>> Also did I also mention that TCP connections don't really fix NAT >>>> issues? >> __________________________________________________________ >> ____________ >>> ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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-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://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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130508/11803035/attachment-0001.html From bedgar at vseinc.com Wed May 8 22:25:28 2013 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Wed, 8 May 2013 14:25:28 -0400 Subject: [Freeswitch-users] Artificial Rings on Inbound TDM calls In-Reply-To: <23DEA328-FE31-4258-84BD-443F52EB6529@freeswitch.org> References: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> <23DEA328-FE31-4258-84BD-443F52EB6529@freeswitch.org> Message-ID: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E6B3@prod-exch01.corp.vseinc.com> Thanks for the reply. I tried ring_ready but that action is after the DNIS and ANI are already received. As ring_ready is part of the dialplan entry for an incoming number that information has to be present before it can get there. I think this would have to be something in the mod_freetdm configurations somewhere. Ideas? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Wednesday, May 8, 2013 1:09 PM To: FreeSWITCH Users Help Cc: Subject: Re: [Freeswitch-users] Artificial Rings on Inbound TDM calls Ring_ready should do that right? Sent from my iPhone On May 8, 2013, at 12:41 PM, > wrote: Hello, Is there any way to present an artificial ring or two to the caller while receiving and processing the DNIS and ANI on an inbound TDM call? There is nearly a 5 second delay during that phase of the call. Thank you, Brian C. Edgar, Jr. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130508/e50f3d9c/attachment.html From msc at freeswitch.org Wed May 8 22:31:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 11:31:02 -0700 Subject: [Freeswitch-users] Artificial Rings on Inbound TDM calls In-Reply-To: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E6B3@prod-exch01.corp.vseinc.com> References: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> <23DEA328-FE31-4258-84BD-443F52EB6529@freeswitch.org> <31A79B0B4414EB4B83C7EAF307ED57DA03B1E6B3@prod-exch01.corp.vseinc.com> Message-ID: Paging Dr. Silva! Dr. Silva to the ML, stat! Yeah, let's ask Moy - he's pretty savvy with that code considering he wrote a ton of it. -MC On Wed, May 8, 2013 at 11:25 AM, wrote: > Thanks for the reply. I tried ring_ready but that action is after the > DNIS and ANI are already received. As ring_ready is part of the dialplan > entry for an incoming number that information has to be present before it > can get there. I think this would have to be something in the mod_freetdm > configurations somewhere.**** > > ** ** > > Ideas?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o > Mesquita > *Sent:* Wednesday, May 8, 2013 1:09 PM > *To:* FreeSWITCH Users Help > *Cc:* > *Subject:* Re: [Freeswitch-users] Artificial Rings on Inbound TDM calls*** > * > > ** ** > > Ring_ready should do that right? > > Sent from my iPhone**** > > > On May 8, 2013, at 12:41 PM, wrote:**** > > Hello,**** > > **** > > Is there any way to present an artificial ring or two to the caller while > receiving and processing the DNIS and ANI on an inbound TDM call? There is > nearly a 5 second delay during that phase of the call.**** > > **** > > Thank you,**** > > **** > > Brian C. Edgar, Jr.**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/d2d7bc1d/attachment.html From bedgar at vseinc.com Wed May 8 22:35:30 2013 From: bedgar at vseinc.com (bedgar at vseinc.com) Date: Wed, 8 May 2013 14:35:30 -0400 Subject: [Freeswitch-users] Artificial Rings on Inbound TDM calls In-Reply-To: References: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E610@prod-exch01.corp.vseinc.com> <23DEA328-FE31-4258-84BD-443F52EB6529@freeswitch.org> <31A79B0B4414EB4B83C7EAF307ED57DA03B1E6B3@prod-exch01.corp.vseinc.com> Message-ID: <31A79B0B4414EB4B83C7EAF307ED57DA03B1E6BE@prod-exch01.corp.vseinc.com> MC, That is my next step as I am currently working with Moises on an effort. I'm sure findings will be posted if a setting exists. BCE From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 8, 2013 2:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Artificial Rings on Inbound TDM calls Paging Dr. Silva! Dr. Silva to the ML, stat! Yeah, let's ask Moy - he's pretty savvy with that code considering he wrote a ton of it. -MC On Wed, May 8, 2013 at 11:25 AM, > wrote: Thanks for the reply. I tried ring_ready but that action is after the DNIS and ANI are already received. As ring_ready is part of the dialplan entry for an incoming number that information has to be present before it can get there. I think this would have to be something in the mod_freetdm configurations somewhere. Ideas? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Wednesday, May 8, 2013 1:09 PM To: FreeSWITCH Users Help Cc: > Subject: Re: [Freeswitch-users] Artificial Rings on Inbound TDM calls Ring_ready should do that right? Sent from my iPhone On May 8, 2013, at 12:41 PM, > wrote: Hello, Is there any way to present an artificial ring or two to the caller while receiving and processing the DNIS and ANI on an inbound TDM call? There is nearly a 5 second delay during that phase of the call. Thank you, Brian C. Edgar, Jr. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/94eeb26e/attachment-0001.html From jeff at askcornerstone.net Wed May 8 22:17:33 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Wed, 8 May 2013 18:17:33 +0000 Subject: [Freeswitch-users] External Softphone vs. Internal Question In-Reply-To: References: <80DFCBDE2AC6574487E3826FAF38F9CC387A7BC3@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387AAC51@vega.terisol.com> <8A9716A5B256904FB1F07C050F9CCCCB020CAB73@mail2.firstdataworks.net> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CAD6A@mail2.firstdataworks.net> No, they're behind different NAT devices. It goes phone > firewall1 > WAN > firewall2 > FS. Ports 5060 and 5090 are forwarded to FS in firewall2 (though 5060 is disabled now for testing). I find in the "domain" setting (on Linphone, "Registrar" on Jisti), I can put anything (1.2.3.4) and it will register as long as the proxy is set to :5090. I've created my external profile according to http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios. Here's the reg info for the profile: Call-ID: 328893541 User: 1000 at 192.168.10.32 Contact: "user" :3773;line=7e848ad901ec2ca;fs_nat=yes;fs_path=sip%3A1000%%3A3773%3Bline%3D7e848ad901ec2ca> Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0) Status: Registered(UDP-NAT)(unknown) EXP(2013-05-08 18:20:07) EXPSECS(524) Host: freeswitch IP: Port: 3773 Auth-User: 1000 Auth-Realm: 1.2.3.4 MWI-Account: 1000 at 192.168.10.32 Total items returned: 1 Here's the status for the ext5090 profile: Name ext5090 Domain Name N/A Auto-NAT false DBName sofia_reg_ext5090 Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.10.32 Ext-RTP-IP SIP-IP 192.168.10.32 Ext-SIP-IP URL sip:mod_sofia@:5090 BIND-URL sip:mod_sofia@:5090;maddr=192.168.10.32;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT true CALLS-IN 30 FAILED-CALLS-IN 4 CALLS-OUT 14 FAILED-CALLS-OUT 4 REGISTRATIONS 1 Thanks, Jeff From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 08, 2013 6:44 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] External Softphone vs. Internal Question On Wed, May 8, 2013 at 2:00 AM, Jeff Bernhardt > wrote: As a follow up to this, I have another question! I got busy with some other things the last couple weeks and was revisiting this again. I realized that even though I thought I was registering to my external5090 profile, I was actually still registering to the internal one on 5060 (5060 is forwarded through my firewall to FS in addition to 5090). I discovered that this is because even though I put what I thought was the registration port 5090 in Jitsi (or Bria, or Linphone, etc.), the softphone was still registering to 5060. Couldn't figure out why, but then I put in the :5090 in the outbound proxy settings, and it worked and confirmed registered to 5090 in fs_cli. Why do I have to put the external ip in the proxy settings on my phone? Something set up wrong in my Freeswitch, my softphone(s)? Confused... Without knowing the topology we are equally confused. Is your phone behind the same NAT device as FreeSWITCH? What happens when you put :5090 in the proxy settings of the phone? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/922af5a2/attachment.html From bellesoft at gmail.com Wed May 8 23:56:15 2013 From: bellesoft at gmail.com (bellesoft) Date: Wed, 8 May 2013 12:56:15 -0700 (PDT) Subject: [Freeswitch-users] server crash with latest git 1.2.stable from today (ab7ae9f) mod_rtmp Message-ID: <1368042975619-7590508.post@n2.nabble.com> Hi, I updated a server to the latest 1.2.stable (ab7ae9f) from earlier today. mod_rtmp seems to be the culprit, each time I disconnect from a flex client, I can reliable crash the server. I am also doing uuid_kill on the channel. I'll get a coredump, and file a JIRA. But wanted to know if anybody else is having the issue. Backtrace here: http://pastebin.freeswitch.org/20887 I also see that there has been a few more commits from the team after, wondering they address this crash? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/server-crash-with-latest-git-1-2-stable-from-today-ab7ae9f-mod-rtmp-tp7590508.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu May 9 00:35:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 08 May 2013 15:35:41 -0500 Subject: [Freeswitch-users] server crash with latest git 1.2.stable from today (ab7ae9f) mod_rtmp In-Reply-To: <1368042975619-7590508.post@n2.nabble.com> Message-ID: Hey open a ticket on this so we can track it... Also try updating to the latest patch set and see if it still happens, if its repeatable on demand let us know that in the ticket. Also please attach the backtrace to the ticket (don't paste it into the comments or link to the pastebin please, both of these things get problematic for the devs working remotely) Thanks K On 5/8/13 2:56 PM, "bellesoft" wrote: > Hi, > I updated a server to the latest 1.2.stable (ab7ae9f) from earlier today. > > mod_rtmp seems to be the culprit, each time I disconnect from a flex client, > I can reliable crash the server. > I am also doing uuid_kill on the channel. > > I'll get a coredump, and file a JIRA. But wanted to know if anybody else is > having the issue. > Backtrace here: http://pastebin.freeswitch.org/20887 > > I also see that there has been a few more commits from the team after, > wondering they address this crash? > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/server-crash-with-latest-git-1-2 > -stable-from-today-ab7ae9f-mod-rtmp-tp7590508.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From anthony.minessale at gmail.com Thu May 9 00:57:24 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 May 2013 15:57:24 -0500 Subject: [Freeswitch-users] pickup endpoint - Break the Simultaneous Ring In-Reply-To: References: Message-ID: Put the pickup in the dial-string for the user instead, you can't use user/xyz in a multi ring bridge line because its already recursive to use the user endpoint. If you add it to the user's dial string it will be fine. you can even make a catch-all one using variables to auto-generate the pickup name so its consistent. You could try using enterprise originate delim instead as well. On Mon, May 6, 2013 at 12:50 PM, Lloyd Aloysius wrote: > Hello All > > The following break the Simultaneous Ring.User registered two different > devices. Only one Device ring after add the *,pickup/ > dave at alcan.mydomain.ca * > > *bridge_data: * > > {sip_invite_domain=alcan.mydomain.ca > ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca > }[leg_timeout=20]user/dave at alcan.mydomain.ca*,pickup/ > dave at alcan.mydomain.ca* > > > > > > Thanks > LLoyd > * * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/afe3efde/attachment.html From drk at drkngs.net Thu May 9 03:06:21 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 08 May 2013 16:06:21 -0700 Subject: [Freeswitch-users] Windows binary MSI installer In-Reply-To: <1368028180930-7590482.post@n2.nabble.com> Message-ID: <20130508230621.d99803bf@mail.tritonwest.net> Did you do both WiX projects the one for 2012 and 2010? --Dave _____ From: Jeff Lenk [mailto:jeff at jefflenk.com] To: freeswitch-users at lists.freeswitch.org Sent: Wed, 08 May 2013 08:49:40 -0700 Subject: Re: [Freeswitch-users] Windows binary MSI installer Ok I reset the build for sound file inclusion so a new msi should be available this weekend. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590482.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://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/20130508/67a2317b/attachment.html From bellesoft at gmail.com Thu May 9 01:42:39 2013 From: bellesoft at gmail.com (bellesoft) Date: Wed, 8 May 2013 14:42:39 -0700 (PDT) Subject: [Freeswitch-users] server crash with latest git 1.2.stable from today (ab7ae9f) mod_rtmp In-Reply-To: <1368042975619-7590508.post@n2.nabble.com> References: <1368042975619-7590508.post@n2.nabble.com> Message-ID: <1368049359366-7590512.post@n2.nabble.com> Thanks Ken, done: http://jira.freeswitch.org/browse/FS-5395 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/server-crash-with-latest-git-1-2-stable-from-today-ab7ae9f-mod-rtmp-tp7590508p7590512.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brandoncoale at sbcglobal.net Thu May 9 04:21:31 2013 From: brandoncoale at sbcglobal.net (Brandon Coale) Date: Wed, 08 May 2013 20:21:31 -0400 Subject: [Freeswitch-users] looking for a way to do appointment reminders Message-ID: <518AEC0B.8070708@sbcglobal.net> Hello, I am in the research phase for the following project and am researching what options are available. My health care organization is looking for a way to do appointment reminders. We currently have staff members who spend part of each day manually calling patients to remind them of their upcoming appointments, and we would like to automate this process. Our electronic health record software would provide such information as the patient's name, phone number, and day and time of the appointment, and FreeSWITCH could take this information and place an automated call to the patient. We would like the reminder call to use text-to-speech to personalize the call, such as "We have an appointment reminder for [first name]. The appointment is on [date] at [time]. I am wondering if anyone has experience with using FreeSWITCH for this type of application, and would be willing to share any details of how you implemented it? I am interested in any ideas, from very simple to feature-rich. We would be doing a new installation of FreeSWITCH for this purpose, so we could use any version of FreeSWITCH or any operating system you would recommend. Thank you! Brandon From jeff at jefflenk.com Thu May 9 06:26:06 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 8 May 2013 19:26:06 -0700 (PDT) Subject: [Freeswitch-users] Windows binary MSI installer In-Reply-To: <20130508230621.d99803bf@mail.tritonwest.net> References: <009101ce4bfd$f6607a10$e3216e30$@v10networks.ca> <1368028180930-7590482.post@n2.nabble.com> <20130508230621.d99803bf@mail.tritonwest.net> Message-ID: <1368066366091-7590514.post@n2.nabble.com> There was no actual code change made. The scripts automatically pickup all the files in the directories but the scripts don't automatically detect new sound file version updates- I just cleaned the sound downloads forcing an update. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Windows-binary-MSI-installer-tp7590459p7590514.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu May 9 06:26:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 19:26:34 -0700 Subject: [Freeswitch-users] External Softphone vs. Internal Question In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CAD6A@mail2.firstdataworks.net> References: <80DFCBDE2AC6574487E3826FAF38F9CC387A7BC3@vega.terisol.com> <80DFCBDE2AC6574487E3826FAF38F9CC387AAC51@vega.terisol.com> <8A9716A5B256904FB1F07C050F9CCCCB020CAB73@mail2.firstdataworks.net> <8A9716A5B256904FB1F07C050F9CCCCB020CAD6A@mail2.firstdataworks.net> Message-ID: That sounds like the way the softphone happens to work. It seems to me that putting :5090 in the proxy field is fine if it indeed works with your softphone + network config + FreeSWITCH config. -MC On Wed, May 8, 2013 at 11:17 AM, Jeff Bernhardt wrote: > No, they?re behind different NAT devices. It goes phone > firewall1 > > WAN > firewall2 > FS. Ports 5060 and 5090 are forwarded to FS in firewall2 > (though 5060 is disabled now for testing). **** > > ** ** > > I find in the ?domain? setting (on Linphone, ?Registrar? on Jisti), I can > put anything (1.2.3.4) and it will register as long as the proxy is set to > :5090. **** > > ** ** > > I?ve created my external profile according to > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios. **** > > ** ** > > Here?s the reg info for the profile:**** > > ** ** > > Call-ID: 328893541**** > > User: 1000 at 192.168.10.32**** > > Contact: "user" :3773;line=7e848ad901ec2ca;fs_nat=yes;fs_path=sip%3A1000%%3A3773%3Bline%3D7e848ad901ec2ca> > **** > > Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)**** > > Status: Registered(UDP-NAT)(unknown) EXP(2013-05-08 > 18:20:07) EXPSECS(524)**** > > Host: freeswitch**** > > IP: **** > > Port: 3773**** > > Auth-User: 1000**** > > Auth-Realm: 1.2.3.4**** > > MWI-Account: 1000 at 192.168.10.32**** > > ** ** > > Total items returned: 1**** > > ** ** > > Here?s the status for the ext5090 profile:**** > > ** ** > > Name ext5090**** > > Domain Name N/A**** > > Auto-NAT false**** > > DBName sofia_reg_ext5090**** > > Pres Hosts **** > > Dialplan XML**** > > Context public**** > > Challenge Realm auto_to**** > > RTP-IP 192.168.10.32**** > > Ext-RTP-IP **** > > SIP-IP 192.168.10.32**** > > Ext-SIP-IP **** > > URL sip:mod_sofia@ > :5090**** > > BIND-URL sip:mod_sofia@ > :5090;maddr=192.168.10.32;transport=udp,tcp**** > > HOLD-MUSIC local_stream://moh**** > > OUTBOUND-PROXY N/A**** > > CODECS IN G722,PCMU,PCMA,GSM**** > > CODECS OUT PCMU,PCMA,GSM**** > > TEL-EVENT 101**** > > DTMF-MODE rfc2833**** > > CNG 13**** > > SESSION-TO 0**** > > MAX-DIALOG 0**** > > NOMEDIA false**** > > LATE-NEG true**** > > PROXY-MEDIA false**** > > ZRTP-PASSTHRU true**** > > AGGRESSIVENAT true**** > > CALLS-IN 30**** > > FAILED-CALLS-IN 4**** > > CALLS-OUT 14**** > > FAILED-CALLS-OUT 4**** > > REGISTRATIONS 1**** > > ** ** > > ** ** > > ** ** > > Thanks,**** > > ** ** > > Jeff **** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, May 08, 2013 6:44 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] External Softphone vs. Internal Question > **** > > ** ** > > ** ** > > ** ** > > On Wed, May 8, 2013 at 2:00 AM, Jeff Bernhardt > wrote:**** > > As a follow up to this, I have another question! I got busy with some > other things the last couple weeks and was revisiting this again. I > realized that even though I thought I was registering to my external5090 > profile, I was actually still registering to the internal one on 5060 (5060 > is forwarded through my firewall to FS in addition to 5090). I discovered > that this is because even though I put what I thought was the registration > port 5090 in Jitsi (or Bria, or Linphone, etc.), the softphone was still > registering to 5060. Couldn?t figure out why, but then I put in the > :5090 in the outbound proxy settings, and it worked and > confirmed registered to 5090 in fs_cli. Why do I have to put the external > ip in the proxy settings on my phone? Something set up wrong in my > Freeswitch, my softphone(s)? Confused?**** > > Without knowing the topology we are equally confused. Is your phone behind > the same NAT device as FreeSWITCH? What happens when you put > :5090 in the proxy settings of the phone?**** > > -MC**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/663a2240/attachment-0001.html From lloyd.aloysius at sunteltech.ca Thu May 9 06:37:09 2013 From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius) Date: Wed, 8 May 2013 22:37:09 -0400 Subject: [Freeswitch-users] pickup endpoint - Break the Simultaneous Ring In-Reply-To: References: Message-ID: Hi Tony Thank you for the kind reply. enterprise originate delim - solve my issue. Thank you for the help Thanks Lloyd * * * * On Wed, May 8, 2013 at 4:57 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Put the pickup in the dial-string for the user instead, you can't use > user/xyz in a multi ring bridge line because its already recursive to use > the user endpoint. If you add it to the user's dial string it will be > fine. you can even make a catch-all one using variables to auto-generate > the pickup name so its consistent. > > You could try using enterprise originate delim instead as well. > > > > On Mon, May 6, 2013 at 12:50 PM, Lloyd Aloysius wrote: > >> Hello All >> >> The following break the Simultaneous Ring.User registered two different >> devices. Only one Device ring after add the *,pickup/ >> dave at alcan.mydomain.ca * >> >> *bridge_data: * >> >> {sip_invite_domain=alcan.mydomain.ca >> ,ignore_early_media=true,force_transfer_context=alcan.mydomain.ca >> }[leg_timeout=20]user/dave at alcan.mydomain.ca*,pickup/ >> dave at alcan.mydomain.ca* >> >> >> >> >> >> Thanks >> LLoyd >> * * >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130508/91c7e49c/attachment.html From msc at freeswitch.org Thu May 9 07:02:30 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 May 2013 20:02:30 -0700 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: <518AEC0B.8070708@sbcglobal.net> References: <518AEC0B.8070708@sbcglobal.net> Message-ID: Hi Brandon, This is totally doable with FreeSWITCH. For you I recommend starting small and working your way up. What I mean by that is that you should probably get the basics down first, like learning how to do a simple dialplan entry in FreeSWITCH, how to set up a SIP provider, how to set up a TTS engine like svox/pico or a commercial one like Cepstral, how to do an originate on the command line and then how to script that using something simple (like fs_cli -x 'originate foo/bar') or more in depth (like ESL)... Then there are the other questions, like: How do you get the patient information out of the software? Is there an API? Will you be keeping track of the results of your automated outbound calls? If so, how will you get the results back into the software? Will you auto-reschedule busy/no answer calls? Do you want to leave a simple message with the called party even if a human answers? Or do you want to have a simple IVR that says something like: "you have an appointment at [date/time]. press 1 to confirm. if you need to reschedule, press 2."? Will you want answering machine detection? If so you'll need to contact FreeSWITCH Solutions about purchasing a license since this is not a free module. Last question: what programming skills do you have on staff there? If you have basic shell scripting skills and/or Perl/PHP/Python/etc. skills then you can probably build something yourself. If you don't have someone with some programming skills then I strongly recommend that you find or hire someone to assist. It's not impossible for a determined person to learn all the skills required, however it will be a challenge. Hope this helps! -Michael P.S. here is a really, really simple example of how to make an outbound call from a shell script: #!/bin/bash # simple dialout script, assumes a gateway named 'flowroute' # and syntax: dialout.sh res=`fs_cli -x '{ignore_early_media=true,fname=$2,lname=$3,appt_timestamp=$4}sofia/gateway/flowroute/$1 reminder` Call that script from shell using a Unix timestamp for the appointment date/time, like this: ./dialout.sh 18005551212 brandon coale 1368077828 Then have a simple dialplan (assumes you've built one of the TTS engines): On Wed, May 8, 2013 at 5:21 PM, Brandon Coale wrote: > Hello, > > I am in the research phase for the following project and am researching > what options are available. > > My health care organization is looking for a way to do appointment > reminders. We currently have staff members who spend part of each day > manually calling patients to remind them of their upcoming appointments, > and we would like to automate this process. > > Our electronic health record software would provide such information as > the patient's name, phone number, and day and time of the appointment, > and FreeSWITCH could take this information and place an automated call > to the patient. We would like the reminder call to use text-to-speech > to personalize the call, such as "We have an appointment reminder for > [first name]. The appointment is on [date] at [time]. > > I am wondering if anyone has experience with using FreeSWITCH for this > type of application, and would be willing to share any details of how > you implemented it? I am interested in any ideas, from very simple to > feature-rich. We would be doing a new installation of FreeSWITCH for > this purpose, so we could use any version of FreeSWITCH or any operating > system you would recommend. > > Thank you! > Brandon > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130508/97c2cc2a/attachment.html From gabe at gundy.org Thu May 9 07:39:20 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 8 May 2013 21:39:20 -0600 Subject: [Freeswitch-users] Freeswitch silent crash In-Reply-To: References: <430001ce41ea$276f6e90$764e4bb0$@ccs.covici.com> <4BF21418-AEAF-4C79-A8C0-28984504B84E@freeswitch.org> <477401ce4499$47347bb0$d59d7310$@gmail.com> Message-ID: On Tue, Apr 30, 2013 at 6:57 AM, Antoine Reversat wrote: > I changed some odbc settings, I'll let you know how it goes. Most of this conversation should have been in Jira, but it wasn't. If the casual reader comes along, he'll see no resolution to this thread. So, I quote from the ticket: """ On the mailing list Anthony asked my to make sure I have the right settings in odbc.ini and odbcinst.ini I was missing those parameters so I added them : UsageCount = 1 FileUsage = 1 Threading = 0 It hasn't crashed since. """ The take away? Always report bugs in Jira & PostgreSQL FTW... OK, maybe not the last part. Gabe From smfarrukh at live.com Thu May 9 07:51:58 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Thu, 9 May 2013 03:51:58 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1368013396577-7590468.post@n2.nabble.com> References: , <1367907318527-7590411.post@n2.nabble.com>, , , , , , <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com>, , , <1368013396577-7590468.post@n2.nabble.com> Message-ID: Hey Mehroz, Did you check that calls are hung up properly, with this setting in vars.xml? or you used X-PRE-PROCESS tag instead? Regards Muhammad Farrukh > Date: Wed, 8 May 2013 04:43:16 -0700 > From: mehroz.ashraf85 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Thanks Minessale , > > Setting up param name="rtp-timeout-sec" value="5" , in vars.xml , solved the > issue. its working flawlessly! > > Thanks for all of yours contributions :) > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590468.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://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/20130509/6d71ce37/attachment.html From dantavious313 at gmail.com Thu May 9 08:08:20 2013 From: dantavious313 at gmail.com (Derrick Dantavious Edwards) Date: Thu, 09 May 2013 00:08:20 -0400 Subject: [Freeswitch-users] mod_python problems Message-ID: <1706053.tWZRyPxI7X@zeus> HI, I am having problems building mod_python. Even though my system python is built with threading support I continue to get the following configure error. Any ideas on what the issue could be? V/r Derrick /configure --enable-64 --enable-zrtp --with-python=/usr/local/bin/python2.7 checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for a thread-safe mkdir -p... build/config/install-sh -c -d checking for gawk... no checking for mawk... no checking for nawk... nawk checking whether make sets $(MAKE)... yes checking build system type... amd64-unknown-freebsd10.0 checking host system type... amd64-unknown-freebsd10.0 checking for gcc... gcc checking whether the C compiler works... yes checking for C compiler default output file name... a.out checking for suffix of executables... checking whether we are cross compiling... no checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ISO C89... none needed checking for style of include used by make... GNU checking dependency style of gcc... gcc3 checking for g++... g++ checking whether we are using the GNU C++ compiler... yes checking whether g++ accepts -g... yes checking dependency style of g++... gcc3 checking whether the C++ compiler works... yes checking for gawk... (cached) nawk checking whether make sets $(MAKE)... (cached) yes checking whether gcc and cc understand -c and -o together... yes checking how to print strings... printf checking for a sed that does not truncate output... /usr/bin/sed checking for grep that handles long lines and -e... /usr/bin/grep checking for egrep... /usr/bin/grep -E checking for fgrep... /usr/bin/grep -F checking for ld used by gcc... /usr/bin/ld checking if the linker (/usr/bin/ld) is GNU ld... yes checking for BSD- or MS-compatible name lister (nm)... /usr/bin/nm -B checking the name lister (/usr/bin/nm -B) interface... BSD nm checking whether ln -s works... yes checking the maximum length of command line arguments... 196608 checking whether the shell understands some XSI constructs... yes checking whether the shell understands "+="... no checking how to convert amd64-unknown-freebsd10.0 file names to amd64-unknown- freebsd10.0 format... func_convert_file_noop checking how to convert amd64-unknown-freebsd10.0 file names to toolchain format... func_convert_file_noop checking for /usr/bin/ld option to reload object files... -r checking for objdump... objdump checking how to recognize dependent libraries... pass_all checking for dlltool... no checking how to associate runtime and link libraries... printf %s\n checking for ar... ar checking for archiver @FILE support... no checking for strip... strip checking for ranlib... ranlib checking command to parse /usr/bin/nm -B output from gcc object... ok checking for sysroot... no checking for mt... mt checking if mt is a manifest tool... no checking how to run the C preprocessor... gcc -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for dlfcn.h... yes checking for objdir... .libs checking if gcc supports -fno-rtti -fno-exceptions... no checking for gcc option to produce PIC... -fPIC -DPIC checking if gcc PIC flag -fPIC -DPIC works... yes checking if gcc static flag -static works... yes checking if gcc supports -c -o file.o... yes checking if gcc supports -c -o file.o... (cached) yes checking whether the gcc linker (/usr/bin/ld) supports shared libraries... yes checking whether -lc should be explicitly linked in... no checking dynamic linker characteristics... freebsd10.0 ld.so checking how to hardcode library paths into programs... immediate checking whether stripping libraries is possible... yes checking if libtool supports shared libraries... yes checking whether to build shared libraries... yes checking whether to build static libraries... yes checking how to run the C++ preprocessor... g++ -E checking for ld used by g++... /usr/bin/ld checking if the linker (/usr/bin/ld) is GNU ld... yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking for g++ option to produce PIC... -fPIC -DPIC checking if g++ PIC flag -fPIC -DPIC works... yes checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whether the g++ linker (/usr/bin/ld) supports shared libraries... yes checking dynamic linker characteristics... freebsd10.0 ld.so checking how to hardcode library paths into programs... immediate checking for C compiler vendor... gnu checking for libgnutls-config... no checking for pkg-config... /usr/local/bin/pkg-config checking for libgnutls - version >= 0.1.0... yes checking libtool major version... 2 using libtool library extension... la adding "-fPIC" to SWITCH_AM_CFLAGS adding "-fPIC" to SWITCH_AM_CXXFLAGS adding "-Werror" to SWITCH_AM_CFLAGS checking whether the compiler supports -fvisibility=hidden... yes adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic adding "-g" to SWITCH_AM_CFLAGS adding "-ggdb" to SWITCH_AM_CFLAGS adding "-DENABLE_ZRTP" to SWITCH_AM_CFLAGS checking for jack... checking for snd_pcm_open in -lasound... no checking size of long... 8 checking what directory libraries are found in... lib checking for odbc header in /usr/include... no found checking for odbc header in /usr/local/include... no found checking for SQLDisconnect in -lodbc... no checking for odbc library in /usr/lib... no found checking for odbc library in /usr/local/lib... no found checking whether to include odbc... no checking for inflateReset in -lz... yes adding "-lz" to SWITCH_AM_LDFLAGS checking for jpeg_std_error in -ljpeg... yes checking for res_init in -lresolv... no adding "-I/usr/local/include" to SWITCH_AM_CFLAGS setting PLATFORM_CORE_LIBS to "-lcrypt -lrt" checking for dirent.h that defines DIR... yes checking for library containing opendir... none required checking for ANSI C header files... (cached) yes checking for sys/types.h... (cached) yes checking sys/resource.h usability... yes checking sys/resource.h presence... yes checking for sys/resource.h... yes checking sched.h usability... yes checking sched.h presence... yes checking for sched.h... yes checking wchar.h usability... yes checking wchar.h presence... yes checking for wchar.h... yes checking sys/filio.h usability... yes checking sys/filio.h presence... yes checking for sys/filio.h... yes checking sys/ioctl.h usability... yes checking sys/ioctl.h presence... yes checking for sys/ioctl.h... yes checking sys/select.h usability... yes checking sys/select.h presence... yes checking for sys/select.h... yes checking netdb.h usability... yes checking netdb.h presence... yes checking for netdb.h... yes checking execinfo.h usability... yes checking execinfo.h presence... yes checking for execinfo.h... yes checking for an ANSI C-conforming const... yes checking for inline... inline checking for size_t... yes checking whether time.h and sys/time.h may both be included... yes checking whether struct tm is in sys/time.h or time.h... time.h checking whether gcc needs -traditional... no checking for stdlib.h... (cached) yes checking for GNU libc compatible malloc... yes checking return type of signal handlers... void checking for strftime... yes checking for gethostname... yes checking for vasprintf... yes checking for mmap... yes checking for mlock... yes checking for mlockall... yes checking for usleep... yes checking for getifaddrs... yes checking for timerfd_create... no checking for getdtablesize... yes checking for posix_openpt... yes checking for sched_setscheduler... yes checking for setpriority... yes checking for setrlimit... yes checking for setgroups... yes checking for initgroups... yes checking for wcsncmp... yes checking for setgroups... (cached) yes checking for asprintf... yes checking for setenv... yes checking for pselect... yes checking for gettimeofday... yes checking for localtime_r... yes checking for gmtime_r... yes checking for strcasecmp... yes checking for stricmp... no checking for _stricmp... no checking whether strerror_r is declared... yes checking for strerror_r... yes checking whether strerror_r returns char *... no checking for sched_setaffinity... no checking for sched_getaffinity... no checking for clock_gettime in -lrt... yes checking for clock_getres in -lrt... yes checking for clock_nanosleep in -lrt... no checking for pthread_setschedparam in -lpthread... yes checking for socket... yes checking for /dev/ptmx... no checking for openpty in -lutil... yes checking for struct tm.tm_gmtoff... yes checking for struct tm.tm_zone... yes checking whether RLIMIT_MEMLOCK is declared... yes checking whether SCHED_RR is declared... yes checking whether SCHED_FIFO is declared... yes checking whether to use mlockall... no, broken for non-root users checking for setenv... (cached) yes checking for strtoll... yes checking for strtoull... yes checking for strtoq... yes checking for strtouq... yes checking for __strtoll... no checking for __strtoull... no checking whether va_list is an array... yes checking whether compiler has __attribute__... yes checking whether compiler supports -Wdeclaration-after-statement... yes yes adding "-Wdeclaration-after-statement" to SWITCH_ANSI_CFLAGS checking whether byte ordering is bigendian... no checking size of char... 1 checking size of int... 4 checking size of long... (cached) 8 checking size of short... 2 checking size of long long... 8 checking for size_t... (cached) yes checking for ssize_t... yes checking size of void*... 8 checking size of ssize_t... 8 checking size of size_t... 8 checking for gunzip... /usr/bin/gunzip checking for bzip2... /usr/bin/bzip2 checking for xz... /usr/bin/xz checking for gtar... no checking for tar... /usr/bin/tar checking for wget... no checking for curl... /usr/local/bin/curl checking whether to use system libcurl library... checking for gawk... (cached) nawk checking for curl-config... /usr/local/bin/curl-config checking for the version of libcurl... 7.24.0 checking for libcurl >= version 7.13.0... yes checking whether libcurl is usable... no no checking for tgetent in -lncurses... yes checking for openssl... checking openssl/tls1.h usability... yes checking openssl/tls1.h presence... yes checking for openssl/tls1.h... yes checking for BIO_new in -lcrypto... yes checking for TLSv1_method in -lssl... yes adding "-DHAVE_OPENSSL" to SWITCH_AM_CFLAGS checking for SSL_CTX_set_tlsext_use_srtp in -lssl... yes checking for DTLSv1_method in -lssl... yes checking for JAVA installation at ... configure: cannot find the java directory, assuming it is specified in CFLAGS checking if JAVA package is complete... no checking for python... /usr/local/bin/python2.7 checking python version... 2.7.3 checking for python distutils... yes checking location of site-packages... /usr/local/lib/python2.7/site-packages checking python libdir... /usr/local/lib checking for main in -lpython2.7... yes checking for PyThread_init_thread... no configure: error: Your python lacks threads support, can not build mod_python /python2.7/dummy_threading.pyc ./python2.7/dummy_thread.pyo ./python2.7/dummy_threading.py ./python2.7/_threading_local.pyo ./python2.7/threading.pyc ./python2.7/_threading_local.py ./python2.7/dummy_thread.pyc ./python2.7/test/test_threadsignals.pyo ./python2.7/test/test_threading.pyc ./python2.7/test/threaded_import_hangers.py ./python2.7/test/threaded_import_hangers.pyc ./python2.7/test/test_threadedtempfile.pyo ./python2.7/test/test_dummy_threading.py ./python2.7/test/test_threaded_import.pyc ./python2.7/test/test_threading_local.pyc ./python2.7/test/test_threaded_import.py ./python2.7/test/test_thread.py ./python2.7/test/test_dummy_thread.pyo ./python2.7/test/test_dummy_threading.pyo ./python2.7/test/test_thread.pyc ./python2.7/test/test_threading.pyo ./python2.7/test/test_threadsignals.pyc ./python2.7/test/test_threadedtempfile.pyc ./python2.7/test/threaded_import_hangers.pyo ./python2.7/test/test_threaded_import.pyo ./python2.7/test/test_threadedtempfile.py ./python2.7/test/test_threading_local.pyo ./python2.7/test/test_dummy_thread.py ./python2.7/test/test_threading.py ./python2.7/test/test_dummy_thread.pyc ./python2.7/test/test_threadsignals.py ./python2.7/test/test_threading_local.py ./python2.7/test/test_dummy_threading.pyc ./python2.7/test/test_thread.pyo ./python2.7/dummy_threading.pyo ./python2.7/dummy_thread.py ./python2.7/_threading_local.pyc ./python2.7/threading.py ./python2.7/threading.pyo ./python2.7/bsddb/test/test_thread.pyo ./python2.7/bsddb/test/test_thread.pyc ./python2.7/bsddb/test/test_thread.py ./python3.3/__pycache__/threading.cpython-33.pyc ./python3.3/__pycache__/_dummy_thread.cpython-33.pyc ./python3.3/__pycache__/_threading_local.cpython-33.pyo ./python3.3/__pycache__/dummy_threading.cpython-33.pyo ./python3.3/__pycache__/threading.cpython-33.pyo ./python3.3/__pycache__/_dummy_thread.cpython-33.pyo ./python3.3/__pycache__/dummy_threading.cpython-33.pyc ./python3.3/__pycache__/_threading_local.cpython-33.pyc ./python3.3/_dummy_thread.py ./python3.3/concurrent/futures/thread.py ./python3.3/concurrent/futures/__pycache__/thread.cpython-33.pyc ./python3.3/concurrent/futures/__pycache__/thread.cpython-33.pyo ./python3.3/test/test_threading_local.py ./python3.3/test/test_threadsignals.py ./python3.3/test/test_dummy_thread.py ./python3.3/test/test_thread.py ./python3.3/test/test_threading.py ./python3.3/test/test_threaded_import.py ./python3.3/test/__pycache__/test_threading.cpython-33.pyc ./python3.3/test/__pycache__/test_threaded_import.cpython-33.pyo ./python3.3/test/__pycache__/test_thread.cpython-33.pyc ./python3.3/test/__pycache__/test_threading_local.cpython-33.pyo ./python3.3/test/__pycache__/test_dummy_threading.cpython-33.pyc ./python3.3/test/__pycache__/test_threadedtempfile.cpython-33.pyc ./python3.3/test/__pycache__/test_threadsignals.cpython-33.pyc ./python3.3/test/__pycache__/threaded_import_hangers.cpython-33.pyc ./python3.3/test/__pycache__/test_dummy_thread.cpython-33.pyo ./python3.3/test/__pycache__/test_threading.cpython-33.pyo ./python3.3/test/__pycache__/test_thread.cpython-33.pyo ./python3.3/test/__pycache__/test_threaded_import.cpython-33.pyc ./python3.3/test/__pycache__/test_threading_local.cpython-33.pyc ./python3.3/test/__pycache__/test_threadedtempfile.cpython-33.pyo ./python3.3/test/__pycache__/test_dummy_threading.cpython-33.pyo ./python3.3/test/__pycache__/threaded_import_hangers.cpython-33.pyo ./python3.3/test/__pycache__/test_threadsignals.cpython-33.pyo ./python3.3/test/__pycache__/test_dummy_thread.cpython-33.pyc ./python3.3/test/test_dummy_threading.py ./python3.3/test/threaded_import_hangers.py ./python3.3/test/test_threadedtempfile.py ./python3.3/dummy_threading.py ./python3.3/_threading_local.py ./python3.3/threading.py ./pth/libpthread.a ./pth/libpthread.so.20 ./pth/libpthread.so From letterstack at gmail.com Thu May 9 08:37:16 2013 From: letterstack at gmail.com (Shiju V.Joseph) Date: Thu, 9 May 2013 10:07:16 +0530 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED Message-ID: HI , sofia_contact can find the user freeswitch at internal> sofia_contact 913288 sofia/internal/sip:913288 at 61.17.226.17:5061 ;line=f3779717978cb25;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 61.17.226.17:36771 ;line=5cd95ba4cffdede;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A36771%3E,sofia/internal/sip:913288 at 61.17.226.17:5061 ;line=6f63c01b0fa6b97;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 124.124.48.35:17394 ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A17394%3E,sofia/internal/sip:913288 at 124.124.48.35:19746 ;line=d43c71dfcd69e5b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19746%3E,sofia/internal/sip:913288 at 124.124.48.35:19747 ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19747%3E but it shows multiple lines , is it normal. Thanks shijujoe On Wed, May 8, 2013 at 3:00 PM, Shiju V.Joseph wrote: > Hi All, > > I have been experimenting with an odbc based freeswitch cluster in amazon > ec2 , with opensips doing the load balancing function. > > I can make calls to mobile and landlines with out any issues with good > quality voice , but when i try to call extension to extension freeswith > shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot > in lists and wiki and fs jira , tried out different dial strings but with > out any results.Tried this one > "dial-string" value="{presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & > this ${sofia_contact(*/${dialed_user}@${dialed_domain})} > > I have copied the siptrace at http://pastebin.com/Qv79tjXK > > Appreciate any help in this regard > > Thanks > -- > shijujoe > -- Shiju V.Joseph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/01bcce68/attachment-0001.html From ashish at nms.co.in Thu May 9 09:57:07 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 9 May 2013 11:27:07 +0530 Subject: [Freeswitch-users] mod_python problems In-Reply-To: <1706053.tWZRyPxI7X@zeus> References: <1706053.tWZRyPxI7X@zeus> Message-ID: You will have to install python with thread support and then install again On Thu, May 9, 2013 at 9:38 AM, Derrick Dantavious Edwards < dantavious313 at gmail.com> wrote: > HI, > I am having problems building mod_python. Even though my system python is > built with threading support I continue to get the following configure > error. > Any ideas on what the issue could be? > V/r > > Derrick > > > /configure --enable-64 --enable-zrtp --with-python=/usr/local/bin/python2.7 > checking for a BSD-compatible install... /usr/bin/install -c > checking whether build environment is sane... yes > checking for a thread-safe mkdir -p... build/config/install-sh -c -d > checking for gawk... no > checking for mawk... no > checking for nawk... nawk > checking whether make sets $(MAKE)... yes > checking build system type... amd64-unknown-freebsd10.0 > checking host system type... amd64-unknown-freebsd10.0 > checking for gcc... gcc > checking whether the C compiler works... yes > checking for C compiler default output file name... a.out > checking for suffix of executables... > checking whether we are cross compiling... no > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether gcc accepts -g... yes > checking for gcc option to accept ISO C89... none needed > checking for style of include used by make... GNU > checking dependency style of gcc... gcc3 > checking for g++... g++ > checking whether we are using the GNU C++ compiler... yes > checking whether g++ accepts -g... yes > checking dependency style of g++... gcc3 > checking whether the C++ compiler works... yes > checking for gawk... (cached) nawk > checking whether make sets $(MAKE)... (cached) yes > checking whether gcc and cc understand -c and -o together... yes > checking how to print strings... printf > checking for a sed that does not truncate output... /usr/bin/sed > checking for grep that handles long lines and -e... /usr/bin/grep > checking for egrep... /usr/bin/grep -E > checking for fgrep... /usr/bin/grep -F > checking for ld used by gcc... /usr/bin/ld > checking if the linker (/usr/bin/ld) is GNU ld... yes > checking for BSD- or MS-compatible name lister (nm)... /usr/bin/nm -B > checking the name lister (/usr/bin/nm -B) interface... BSD nm > checking whether ln -s works... yes > checking the maximum length of command line arguments... 196608 > checking whether the shell understands some XSI constructs... yes > checking whether the shell understands "+="... no > checking how to convert amd64-unknown-freebsd10.0 file names to > amd64-unknown- > freebsd10.0 format... func_convert_file_noop > checking how to convert amd64-unknown-freebsd10.0 file names to toolchain > format... func_convert_file_noop > checking for /usr/bin/ld option to reload object files... -r > checking for objdump... objdump > checking how to recognize dependent libraries... pass_all > checking for dlltool... no > checking how to associate runtime and link libraries... printf %s\n > checking for ar... ar > checking for archiver @FILE support... no > checking for strip... strip > checking for ranlib... ranlib > checking command to parse /usr/bin/nm -B output from gcc object... ok > checking for sysroot... no > checking for mt... mt > checking if mt is a manifest tool... no > checking how to run the C preprocessor... gcc -E > checking for ANSI C header files... yes > checking for sys/types.h... yes > checking for sys/stat.h... yes > checking for stdlib.h... yes > checking for string.h... yes > checking for memory.h... yes > checking for strings.h... yes > checking for inttypes.h... yes > checking for stdint.h... yes > checking for unistd.h... yes > checking for dlfcn.h... yes > checking for objdir... .libs > checking if gcc supports -fno-rtti -fno-exceptions... no > checking for gcc option to produce PIC... -fPIC -DPIC > checking if gcc PIC flag -fPIC -DPIC works... yes > checking if gcc static flag -static works... yes > checking if gcc supports -c -o file.o... yes > checking if gcc supports -c -o file.o... (cached) yes > checking whether the gcc linker (/usr/bin/ld) supports shared libraries... > yes > checking whether -lc should be explicitly linked in... no > checking dynamic linker characteristics... freebsd10.0 ld.so > checking how to hardcode library paths into programs... immediate > checking whether stripping libraries is possible... yes > checking if libtool supports shared libraries... yes > checking whether to build shared libraries... yes > checking whether to build static libraries... yes > checking how to run the C++ preprocessor... g++ -E > checking for ld used by g++... /usr/bin/ld > checking if the linker (/usr/bin/ld) is GNU ld... yes > checking whether the g++ linker (/usr/bin/ld) supports shared libraries... > yes > checking for g++ option to produce PIC... -fPIC -DPIC > checking if g++ PIC flag -fPIC -DPIC works... yes > checking if g++ static flag -static works... yes > checking if g++ supports -c -o file.o... yes > checking if g++ supports -c -o file.o... (cached) yes > checking whether the g++ linker (/usr/bin/ld) supports shared libraries... > yes > checking dynamic linker characteristics... freebsd10.0 ld.so > checking how to hardcode library paths into programs... immediate > checking for C compiler vendor... gnu > checking for libgnutls-config... no > checking for pkg-config... /usr/local/bin/pkg-config > checking for libgnutls - version >= 0.1.0... yes > checking libtool major version... 2 > using libtool library extension... la > adding "-fPIC" to SWITCH_AM_CFLAGS > adding "-fPIC" to SWITCH_AM_CXXFLAGS > adding "-Werror" to SWITCH_AM_CFLAGS > checking whether the compiler supports -fvisibility=hidden... yes > adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS > adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS > adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS > adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS > adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS > adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS > checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic > adding "-g" to SWITCH_AM_CFLAGS > adding "-ggdb" to SWITCH_AM_CFLAGS > adding "-DENABLE_ZRTP" to SWITCH_AM_CFLAGS > checking for jack... checking for snd_pcm_open in -lasound... no > checking size of long... 8 > checking what directory libraries are found in... lib > checking for odbc header in /usr/include... no found > checking for odbc header in /usr/local/include... no found > checking for SQLDisconnect in -lodbc... no > checking for odbc library in /usr/lib... no found > checking for odbc library in /usr/local/lib... no found > checking whether to include odbc... no > checking for inflateReset in -lz... yes > adding "-lz" to SWITCH_AM_LDFLAGS > checking for jpeg_std_error in -ljpeg... yes > checking for res_init in -lresolv... no > adding "-I/usr/local/include" to SWITCH_AM_CFLAGS > setting PLATFORM_CORE_LIBS to "-lcrypt -lrt" > checking for dirent.h that defines DIR... yes > checking for library containing opendir... none required > checking for ANSI C header files... (cached) yes > checking for sys/types.h... (cached) yes > checking sys/resource.h usability... yes > checking sys/resource.h presence... yes > checking for sys/resource.h... yes > checking sched.h usability... yes > checking sched.h presence... yes > checking for sched.h... yes > checking wchar.h usability... yes > checking wchar.h presence... yes > checking for wchar.h... yes > checking sys/filio.h usability... yes > checking sys/filio.h presence... yes > checking for sys/filio.h... yes > checking sys/ioctl.h usability... yes > checking sys/ioctl.h presence... yes > checking for sys/ioctl.h... yes > checking sys/select.h usability... yes > checking sys/select.h presence... yes > checking for sys/select.h... yes > checking netdb.h usability... yes > checking netdb.h presence... yes > checking for netdb.h... yes > checking execinfo.h usability... yes > checking execinfo.h presence... yes > checking for execinfo.h... yes > checking for an ANSI C-conforming const... yes > checking for inline... inline > checking for size_t... yes > checking whether time.h and sys/time.h may both be included... yes > checking whether struct tm is in sys/time.h or time.h... time.h > checking whether gcc needs -traditional... no > checking for stdlib.h... (cached) yes > checking for GNU libc compatible malloc... yes > checking return type of signal handlers... void > checking for strftime... yes > checking for gethostname... yes > checking for vasprintf... yes > checking for mmap... yes > checking for mlock... yes > checking for mlockall... yes > checking for usleep... yes > checking for getifaddrs... yes > checking for timerfd_create... no > checking for getdtablesize... yes > checking for posix_openpt... yes > checking for sched_setscheduler... yes > checking for setpriority... yes > checking for setrlimit... yes > checking for setgroups... yes > checking for initgroups... yes > checking for wcsncmp... yes > checking for setgroups... (cached) yes > checking for asprintf... yes > checking for setenv... yes > checking for pselect... yes > checking for gettimeofday... yes > checking for localtime_r... yes > checking for gmtime_r... yes > checking for strcasecmp... yes > checking for stricmp... no > checking for _stricmp... no > checking whether strerror_r is declared... yes > checking for strerror_r... yes > checking whether strerror_r returns char *... no > checking for sched_setaffinity... no > checking for sched_getaffinity... no > checking for clock_gettime in -lrt... yes > checking for clock_getres in -lrt... yes > checking for clock_nanosleep in -lrt... no > checking for pthread_setschedparam in -lpthread... yes > checking for socket... yes > checking for /dev/ptmx... no > checking for openpty in -lutil... yes > checking for struct tm.tm_gmtoff... yes > checking for struct tm.tm_zone... yes > checking whether RLIMIT_MEMLOCK is declared... yes > checking whether SCHED_RR is declared... yes > checking whether SCHED_FIFO is declared... yes > checking whether to use mlockall... no, broken for non-root users > checking for setenv... (cached) yes > checking for strtoll... yes > checking for strtoull... yes > checking for strtoq... yes > checking for strtouq... yes > checking for __strtoll... no > checking for __strtoull... no > checking whether va_list is an array... yes > checking whether compiler has __attribute__... yes > checking whether compiler supports -Wdeclaration-after-statement... yes > yes > adding "-Wdeclaration-after-statement" to SWITCH_ANSI_CFLAGS > checking whether byte ordering is bigendian... no > checking size of char... 1 > checking size of int... 4 > checking size of long... (cached) 8 > checking size of short... 2 > checking size of long long... 8 > checking for size_t... (cached) yes > checking for ssize_t... yes > checking size of void*... 8 > checking size of ssize_t... 8 > checking size of size_t... 8 > checking for gunzip... /usr/bin/gunzip > checking for bzip2... /usr/bin/bzip2 > checking for xz... /usr/bin/xz > checking for gtar... no > checking for tar... /usr/bin/tar > checking for wget... no > checking for curl... /usr/local/bin/curl > checking whether to use system libcurl library... checking for gawk... > (cached) nawk > checking for curl-config... /usr/local/bin/curl-config > checking for the version of libcurl... 7.24.0 > checking for libcurl >= version 7.13.0... yes > checking whether libcurl is usable... no > no > checking for tgetent in -lncurses... yes > checking for openssl... checking openssl/tls1.h usability... yes > checking openssl/tls1.h presence... yes > checking for openssl/tls1.h... yes > checking for BIO_new in -lcrypto... yes > checking for TLSv1_method in -lssl... yes > adding "-DHAVE_OPENSSL" to SWITCH_AM_CFLAGS > checking for SSL_CTX_set_tlsext_use_srtp in -lssl... yes > checking for DTLSv1_method in -lssl... yes > checking for JAVA installation at ... > configure: cannot find the java directory, assuming it is specified in > CFLAGS > checking if JAVA package is complete... no > checking for python... /usr/local/bin/python2.7 > checking python version... 2.7.3 > checking for python distutils... yes > checking location of site-packages... > /usr/local/lib/python2.7/site-packages > checking python libdir... /usr/local/lib > checking for main in -lpython2.7... yes > checking for PyThread_init_thread... no > configure: error: Your python lacks threads support, can not build > mod_python > > > > /python2.7/dummy_threading.pyc > ./python2.7/dummy_thread.pyo > ./python2.7/dummy_threading.py > ./python2.7/_threading_local.pyo > ./python2.7/threading.pyc > ./python2.7/_threading_local.py > ./python2.7/dummy_thread.pyc > ./python2.7/test/test_threadsignals.pyo > ./python2.7/test/test_threading.pyc > ./python2.7/test/threaded_import_hangers.py > ./python2.7/test/threaded_import_hangers.pyc > ./python2.7/test/test_threadedtempfile.pyo > ./python2.7/test/test_dummy_threading.py > ./python2.7/test/test_threaded_import.pyc > ./python2.7/test/test_threading_local.pyc > ./python2.7/test/test_threaded_import.py > ./python2.7/test/test_thread.py > ./python2.7/test/test_dummy_thread.pyo > ./python2.7/test/test_dummy_threading.pyo > ./python2.7/test/test_thread.pyc > ./python2.7/test/test_threading.pyo > ./python2.7/test/test_threadsignals.pyc > ./python2.7/test/test_threadedtempfile.pyc > ./python2.7/test/threaded_import_hangers.pyo > ./python2.7/test/test_threaded_import.pyo > ./python2.7/test/test_threadedtempfile.py > ./python2.7/test/test_threading_local.pyo > ./python2.7/test/test_dummy_thread.py > ./python2.7/test/test_threading.py > ./python2.7/test/test_dummy_thread.pyc > ./python2.7/test/test_threadsignals.py > ./python2.7/test/test_threading_local.py > ./python2.7/test/test_dummy_threading.pyc > ./python2.7/test/test_thread.pyo > ./python2.7/dummy_threading.pyo > ./python2.7/dummy_thread.py > ./python2.7/_threading_local.pyc > ./python2.7/threading.py > ./python2.7/threading.pyo > ./python2.7/bsddb/test/test_thread.pyo > ./python2.7/bsddb/test/test_thread.pyc > ./python2.7/bsddb/test/test_thread.py > > ./python3.3/__pycache__/threading.cpython-33.pyc > ./python3.3/__pycache__/_dummy_thread.cpython-33.pyc > ./python3.3/__pycache__/_threading_local.cpython-33.pyo > ./python3.3/__pycache__/dummy_threading.cpython-33.pyo > ./python3.3/__pycache__/threading.cpython-33.pyo > ./python3.3/__pycache__/_dummy_thread.cpython-33.pyo > ./python3.3/__pycache__/dummy_threading.cpython-33.pyc > ./python3.3/__pycache__/_threading_local.cpython-33.pyc > ./python3.3/_dummy_thread.py > ./python3.3/concurrent/futures/thread.py > ./python3.3/concurrent/futures/__pycache__/thread.cpython-33.pyc > ./python3.3/concurrent/futures/__pycache__/thread.cpython-33.pyo > ./python3.3/test/test_threading_local.py > ./python3.3/test/test_threadsignals.py > ./python3.3/test/test_dummy_thread.py > ./python3.3/test/test_thread.py > ./python3.3/test/test_threading.py > ./python3.3/test/test_threaded_import.py > ./python3.3/test/__pycache__/test_threading.cpython-33.pyc > ./python3.3/test/__pycache__/test_threaded_import.cpython-33.pyo > ./python3.3/test/__pycache__/test_thread.cpython-33.pyc > ./python3.3/test/__pycache__/test_threading_local.cpython-33.pyo > ./python3.3/test/__pycache__/test_dummy_threading.cpython-33.pyc > ./python3.3/test/__pycache__/test_threadedtempfile.cpython-33.pyc > ./python3.3/test/__pycache__/test_threadsignals.cpython-33.pyc > ./python3.3/test/__pycache__/threaded_import_hangers.cpython-33.pyc > ./python3.3/test/__pycache__/test_dummy_thread.cpython-33.pyo > ./python3.3/test/__pycache__/test_threading.cpython-33.pyo > ./python3.3/test/__pycache__/test_thread.cpython-33.pyo > ./python3.3/test/__pycache__/test_threaded_import.cpython-33.pyc > ./python3.3/test/__pycache__/test_threading_local.cpython-33.pyc > ./python3.3/test/__pycache__/test_threadedtempfile.cpython-33.pyo > ./python3.3/test/__pycache__/test_dummy_threading.cpython-33.pyo > ./python3.3/test/__pycache__/threaded_import_hangers.cpython-33.pyo > ./python3.3/test/__pycache__/test_threadsignals.cpython-33.pyo > ./python3.3/test/__pycache__/test_dummy_thread.cpython-33.pyc > ./python3.3/test/test_dummy_threading.py > ./python3.3/test/threaded_import_hangers.py > ./python3.3/test/test_threadedtempfile.py > ./python3.3/dummy_threading.py > ./python3.3/_threading_local.py > ./python3.3/threading.py > > ./pth/libpthread.a > ./pth/libpthread.so.20 > ./pth/libpthread.so > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/dfd62ad2/attachment-0001.html From mehroz.ashraf85 at gmail.com Thu May 9 11:29:48 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Thu, 9 May 2013 00:29:48 -0700 (PDT) Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> <1368013396577-7590468.post@n2.nabble.com> Message-ID: <1368084588897-7590524.post@n2.nabble.com> Yes not in X-PRE-PROCESS, but with param name="rtp-timeout-sec" value="5" in vars.xml. I tested the scenario switching off the Wifi of both client.. it worked fine and call was terminated exactly after 5 sec. But today, i have seen another scenario, i.e, when one of the sip mobile app crashes down, it does not terminates that call. Either RTP is still in flow or something is misunderstood, However i can see the crashed app SIP device gone unregistered on freeswitch after ping option fails. !!!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590524.html Sent from the freeswitch-users mailing list archive at Nabble.com. From smfarrukh at live.com Thu May 9 11:43:30 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Thu, 9 May 2013 07:43:30 +0000 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1368084588897-7590524.post@n2.nabble.com> References: , , , , , <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com>, , , <1368013396577-7590468.post@n2.nabble.com>, , <1368084588897-7590524.post@n2.nabble.com> Message-ID: Hey Mehroz, I checked it with on my Android phone, it hangs up call when I go outside the range of Wifi and connect to Wifi again, and for the duration I am disconnected to Wifi RTP traffic doesn't stop from FS, I tested it in conference call (3100), > Date: Thu, 9 May 2013 00:29:48 -0700 > From: mehroz.ashraf85 at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Dead channels in freeswitch > > Yes not in X-PRE-PROCESS, but with > param name="rtp-timeout-sec" value="5" in vars.xml. > > I tested the scenario switching off the Wifi of both client.. it worked fine > and call was terminated exactly after 5 sec. > > But today, i have seen another scenario, i.e, when one of the sip mobile app > crashes down, it does not terminates that call. Either RTP is still in flow > or something is misunderstood, However i can see the crashed app SIP device > gone unregistered on freeswitch after ping option fails. !!!! > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590524.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://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/20130509/a3d2b7a6/attachment.html From andrew at cassidywebservices.co.uk Thu May 9 12:45:00 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 9 May 2013 09:45:00 +0100 Subject: [Freeswitch-users] TCP vs UDP SIP In-Reply-To: <06BFB360-C378-46AE-B894-C7552966B241@5ninesolutions.com> References: <2d8cabcf-f976-48f4-ae23-26a246df2aea@blur> <00bd01ce4c02$02281570$06784050$@v10networks.ca> <001201ce4c0c$3a485320$aed8f960$@v10networks.ca> <2BF00B16-7505-4D5B-8E83-24AE07BA857D@5ninesolutions.com> <000601ce4c10$5fad6570$1f083050$@v10networks.ca> <06BFB360-C378-46AE-B894-C7552966B241@5ninesolutions.com> Message-ID: I've come across difficulties with TCP and NAT. Some NAT implementations do not allow new connections from the PBX to the phone through the NAT and there's no requirement to reuse and existing, long-lived TCP connection. Some notifications and invites can get blocked unnecessarily. On 8 May 2013 19:13, Spencer Thomason wrote: > Understood. My plan is to use UDP for all "trunking" type endpoints at > TCP for desk phones as they will likely receive more NOTIFYs and in most > cases being behind NAT where the longer connection timeout comes in handy. > I also found a good paper on the subject: > http://www.cs.columbia.edu/~kumiko/publish/IPTComm08_paper.pdf > > In regard to connection timeout how does Freeswitch handle this? I > noticed the new Sofia parameters and I was curious if the connection > lifetime was configurable as well. > > BR, > Spencer > > On May 8, 2013, at 10:20 AM, "Jeff Leung" wrote: > > Hi Jeff, > > Thanks for the insight. Forgive my ignorance but if I have two Identical > > Freeswitch servers with SRV records and endpoints that properly support > > SRVs, why do I loose the ability to failover if one host is not reachable? > > > TCP is a stateful protocol. On the other hand UDP isn't, it's stateless. > It's just easier to failover with UDP than with TCP if you understand the > difference between the two protocols. I'm not saying that it's not possible > to do so with TCP, but with the way how SIP works, you'd want to use UDP if > you want failover capabilities without the headache. > > Also as many of these end points are Polycoms behind NAT, I can't see any > > reason I'd still need NDLB-force-rport on the profile? > > > > Unfortunately, I don't work with Polycom phones. Brian West over here can > comment on that issue. > > Since these are application servers, handling conferences, presence, etc., > > I'd > > imagine I would hit other bottlenecks before I hit the TCP connection > > limit. > > Yes that's true, but if you had a FreeSWITCH box that purely handled SIP > messages and no media, you'd probably hit that TCP Open connection limit. > > On May 8, 2013, at 9:51 AM, "Jeff Leung" wrote: > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of > > Vik Killa > > Sent: Wednesday, May 8, 2013 9:18 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] TCP vs UDP SIP > > > That I would agree with, but the thing is you lose the capability of > > failover in the unlikely event that a node in a FreeSWITCH cluster fail. > > > In my opinion, TCP seems better than UDP as you know all the SIP > > packets are making to their destination. > > > On Wed, May 8, 2013 at 11:37 AM, Jeff Leung > > wrote: > > > > On a Linux system there is a limit of how many open TCP > > connections you have. > > > If I can remember correctly, I think Darren from 2600hz did discuss > > about the limit of open TCP connections you can have on a Linux > > system. Correct me if I'm wrong on this, but that seems to be the > > case. And I have seen instances of that happening on a misconfigured > > Squid Proxy > > > I never heard this before...where and how it this limit defined? > > > > Unless you have a crazy amount of endpoints you have to serve, TCP > > probably isn't really worth it in my opinion. > > > Assuming it's one Open TCP connection per endpoint, you'd probably > > need more endpoints than the maximum amount of open TCP connections > > to > > hit that problem > > > How many endpoints? > > > > Also did I also mention that TCP connections don't really fix NAT > > issues? > > > > > > __________________________________________________________ > > ____________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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-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://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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *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/20130509/9a6bf687/attachment-0001.html From steveayre at gmail.com Thu May 9 13:01:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 9 May 2013 10:01:21 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: <1368084588897-7590524.post@n2.nabble.com> References: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> <1368013396577-7590468.post@n2.nabble.com> <1368084588897-7590524.post@n2.nabble.com> Message-ID: No, it'll be within the tag of the sofia . If you put it in vars.xml it'll be in completely the wrong place and mod_sofia won't see it... On 9 May 2013 08:29, mehroz wrote: > Yes not in X-PRE-PROCESS, but with > param name="rtp-timeout-sec" value="5" in vars.xml. > > I tested the scenario switching off the Wifi of both client.. it worked fine > and call was terminated exactly after 5 sec. > > But today, i have seen another scenario, i.e, when one of the sip mobile app > crashes down, it does not terminates that call. Either RTP is still in flow > or something is misunderstood, However i can see the crashed app SIP device > gone unregistered on freeswitch after ping option fails. !!!! > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590524.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://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 From covici at ccs.covici.com Thu May 9 13:55:41 2013 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 09 May 2013 05:55:41 -0400 Subject: [Freeswitch-users] odcb parameters (was Freeswitch silent crash) In-Reply-To: References: <430001ce41ea$276f6e90$764e4bb0$@ccs.covici.com> <4BF21418-AEAF-4C79-A8C0-28984504B84E@freeswitch.org> <477401ce4499$47347bb0$d59d7310$@gmail.com> Message-ID: <17791.1368093341@ccs.covici.com> I have not heard of usagecount or fileusage where do they go and what do they do? Gabriel Gunderson wrote: > On Tue, Apr 30, 2013 at 6:57 AM, Antoine Reversat wrote: > > I changed some odbc settings, I'll let you know how it goes. > > Most of this conversation should have been in Jira, but it wasn't. If > the casual reader comes along, he'll see no resolution to this thread. > So, I quote from the ticket: > > """ > On the mailing list Anthony asked my to make sure I have the right > settings in odbc.ini and odbcinst.ini I was missing those parameters > so I added them : > > UsageCount = 1 > FileUsage = 1 > Threading = 0 > > It hasn't crashed since. > """ > > The take away? Always report bugs in Jira & PostgreSQL FTW... OK, > maybe not the last part. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- 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 bdfoster at davri.com Thu May 9 14:43:14 2013 From: bdfoster at davri.com (Brian Foster) Date: Thu, 9 May 2013 06:43:14 -0400 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: *Sidebar* Possible recordings for the next session with Callie? Like: 1. "This is an important reminder for..." 2. "This is a reminder for your appointment scheduled for..." ...or something similar. I think it would get some usage with those in the medical field. On May 8, 2013 11:09 PM, "Michael Collins" wrote: > > Hi Brandon, > > This is totally doable with FreeSWITCH. For you I recommend starting small and working your way up. What I mean by that is that you should probably get the basics down first, like learning how to do a simple dialplan entry in FreeSWITCH, how to set up a SIP provider, how to set up a TTS engine like svox/pico or a commercial one like Cepstral, how to do an originate on the command line and then how to script that using something simple (like fs_cli -x 'originate foo/bar') or more in depth (like ESL)... > > Then there are the other questions, like: > How do you get the patient information out of the software? Is there an API? > Will you be keeping track of the results of your automated outbound calls? > If so, how will you get the results back into the software? Will you auto-reschedule busy/no answer calls? > Do you want to leave a simple message with the called party even if a human answers? > Or do you want to have a simple IVR that says something like: "you have an appointment at [date/time]. press 1 to confirm. if you need to reschedule, press 2."? > Will you want answering machine detection? If so you'll need to contact FreeSWITCH Solutions about purchasing a license since this is not a free module. > > Last question: what programming skills do you have on staff there? If you have basic shell scripting skills and/or Perl/PHP/Python/etc. skills then you can probably build something yourself. If you don't have someone with some programming skills then I strongly recommend that you find or hire someone to assist. It's not impossible for a determined person to learn all the skills required, however it will be a challenge. > > Hope this helps! > -Michael > > P.S. here is a really, really simple example of how to make an outbound call from a shell script: > > #!/bin/bash > # simple dialout script, assumes a gateway named 'flowroute' > # and syntax: dialout.sh > res=`fs_cli -x '{ignore_early_media=true,fname=$2,lname=$3,appt_timestamp=$4}sofia/gateway/flowroute/$1 reminder` > > Call that script from shell using a Unix timestamp for the appointment date/time, like this: > ./dialout.sh 18005551212 brandon coale 1368077828 > > Then have a simple dialplan (assumes you've built one of the TTS engines): > > > > > > > > > > > > > > > > On Wed, May 8, 2013 at 5:21 PM, Brandon Coale wrote: >> >> Hello, >> >> I am in the research phase for the following project and am researching >> what options are available. >> >> My health care organization is looking for a way to do appointment >> reminders. We currently have staff members who spend part of each day >> manually calling patients to remind them of their upcoming appointments, >> and we would like to automate this process. >> >> Our electronic health record software would provide such information as >> the patient's name, phone number, and day and time of the appointment, >> and FreeSWITCH could take this information and place an automated call >> to the patient. We would like the reminder call to use text-to-speech >> to personalize the call, such as "We have an appointment reminder for >> [first name]. The appointment is on [date] at [time]. >> >> I am wondering if anyone has experience with using FreeSWITCH for this >> type of application, and would be willing to share any details of how >> you implemented it? I am interested in any ideas, from very simple to >> feature-rich. We would be doing a new installation of FreeSWITCH for >> this purpose, so we could use any version of FreeSWITCH or any operating >> system you would recommend. >> >> Thank you! >> Brandon >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/0aae8237/attachment.html From avi at avimarcus.net Thu May 9 14:49:39 2013 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 9 May 2013 13:49:39 +0300 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: Medical isn't the only industry with appointments. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/9a15c23f/attachment-0001.html From ashish at nms.co.in Thu May 9 15:49:07 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 9 May 2013 17:19:07 +0530 Subject: [Freeswitch-users] mod_perl not loading Message-ID: Hi, I am getting this error while reloading the mod_perl: [CRIT] switch_loadable_module.c:1504 Module is not unloadable. Please help. -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/38d6d215/attachment.html From steveayre at gmail.com Thu May 9 16:08:35 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 9 May 2013 13:08:35 +0100 Subject: [Freeswitch-users] mod_perl not loading In-Reply-To: References: Message-ID: <3DE6DECF-F01F-4BEE-BDE3-B25D875A1569@gmail.com> The module is already loaded. It can't be reloaded. If you really need to that requires restarting FS for that module. What are you trying to do that would require reloading that module? Perhaps you don't actually need to? Steve On 9 May 2013, at 12:49, Ashish gautam wrote: > Hi, > > I am getting this error while reloading the mod_perl: > > [CRIT] switch_loadable_module.c:1504 Module is not unloadable. > > Please help. > > -Ashish > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ashish at nms.co.in Thu May 9 16:22:25 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 9 May 2013 17:52:25 +0530 Subject: [Freeswitch-users] mod_perl not loading In-Reply-To: <3DE6DECF-F01F-4BEE-BDE3-B25D875A1569@gmail.com> References: <3DE6DECF-F01F-4BEE-BDE3-B25D875A1569@gmail.com> Message-ID: Actually I have a perl script that I want to execute from my dialplan. When that script is about to load it shows this error: 2013-05-09 17:51:53.125026 [ERR] mod_perl.c:72 [require '/usr/local/freeswitch/scripts/perl/ash.pl';] Can't locate /usr/local/freeswitch/scripts/perl/ash.pl in @INC (@INC contains: /usr/local/freeswitch/perl /usr/local/lib/perl5/5.14.2/BSDPAN /usr/local/lib/perl5/site_perl/5.14.2/mach /usr/local/lib/perl5/site_perl/5.14.2 /usr/local/lib/perl5/5.14.2/mach /usr/local/lib/perl5/5.14.2 .) at (eval 4) line 1. What is the problem?. On Thu, May 9, 2013 at 5:38 PM, Steven Ayre wrote: > The module is already loaded. It can't be reloaded. If you really need to > that requires restarting FS for that module. > > What are you trying to do that would require reloading that module? > Perhaps you don't actually need to? > > Steve > > On 9 May 2013, at 12:49, Ashish gautam wrote: > > > Hi, > > > > I am getting this error while reloading the mod_perl: > > > > [CRIT] switch_loadable_module.c:1504 Module is not unloadable. > > > > Please help. > > > > -Ashish > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/fc4e1e9c/attachment.html From steveayre at gmail.com Thu May 9 17:26:04 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 9 May 2013 14:26:04 +0100 Subject: [Freeswitch-users] mod_perl not loading In-Reply-To: References: <3DE6DECF-F01F-4BEE-BDE3-B25D875A1569@gmail.com> Message-ID: The file you're trying to run doesn't exist in any of the locations it lists. What's the , and what is the full path of the file? -Steve On 9 May 2013 13:22, Ashish gautam wrote: > Actually I have a perl script that I want to execute from my dialplan. When > that script is about to load it shows this error: > > 2013-05-09 17:51:53.125026 [ERR] mod_perl.c:72 [require > '/usr/local/freeswitch/scripts/perl/ash.pl';] > Can't locate /usr/local/freeswitch/scripts/perl/ash.pl in @INC (@INC > contains: /usr/local/freeswitch/perl /usr/local/lib/perl5/5.14.2/BSDPAN > /usr/local/lib/perl5/site_perl/5.14.2/mach > /usr/local/lib/perl5/site_perl/5.14.2 /usr/local/lib/perl5/5.14.2/mach > /usr/local/lib/perl5/5.14.2 .) at (eval 4) line 1. > > What is the problem?. > > On Thu, May 9, 2013 at 5:38 PM, Steven Ayre wrote: >> >> The module is already loaded. It can't be reloaded. If you really need to >> that requires restarting FS for that module. >> >> What are you trying to do that would require reloading that module? >> Perhaps you don't actually need to? >> >> Steve >> >> On 9 May 2013, at 12:49, Ashish gautam wrote: >> >> > Hi, >> > >> > I am getting this error while reloading the mod_perl: >> > >> > [CRIT] switch_loadable_module.c:1504 Module is not unloadable. >> > >> > Please help. >> > >> > -Ashish >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From mogsy.uk at gmail.com Thu May 9 17:32:42 2013 From: mogsy.uk at gmail.com (Rob Moore) Date: Thu, 9 May 2013 14:32:42 +0100 Subject: [Freeswitch-users] Call Hangup cause 'None' In-Reply-To: References: Message-ID: Bump, anyone there ? On Tue, May 7, 2013 at 1:11 PM, Rob Moore wrote: > Hi Guys, I was wondering if someone could clear up exactly what the hangup > case 'none' means? It started appearing after an upgrade to the latest > version of Freeswitch (from a very very old 1.0.0 version) in place of the > usual "ALOTTED_TIMEOUT" outcomes we would expect to see from and unanswered > call. > > I would have considered this a simple reclassification of the call result > however we are still getting a few "ALOTTED_TIMEOUT" mixed in with these > "none's" which makes me wonder if its actually the symptom of another > problem I'm not seeing yet. > > Any advice on what none means and why it appears in our CDRS would be > greatly appreciated! > > Thanks > > Rob > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/41fbde1b/attachment.html From xiaofengcanyuexp at 163.com Thu May 9 18:36:32 2013 From: xiaofengcanyuexp at 163.com (=?utf-8?B?eGlhb2ZlbmdjYW55dWV4cEAxNjMuY29t?=) Date: Thu, 9 May 2013 22:36:32 +0800 Subject: [Freeswitch-users] =?utf-8?q?Does_freeswitch_support_SIP-T/SIP-T?= =?utf-8?q?=3F?= References: <201303242132282818393@163.com>, <201303242142599068438@163.com>, Message-ID: <201305092236322508083@163.com> Hi, Moises, ?? As you suggested, I have completed the ISUP<->SIP-T conversion work. While there are some problems to integrate my code to freeswitch. 1. I got one sangoma license and can bridge the freetmd to sofia now. But how can I intercept the ISUP mesasge(from the freetdm) and run to my application and then to Sofia module? Is there any external control like ESL to do the work? The diagram is like: SIP-T ISUP SofiaModule<-------------My Conversion---------------------->freetdm 2. I have been working with ESL "SEND" command to control the Sofia module, while it needs to divide the conversion SIP-T message. Is there anyway to put the one whole SIP-T message to Sofia module to handle? like one external SIP-T message to freeswitch? And vice versa, Is there good way to receive the whole SIP message from sofia module instead of using ESL to detect the SIP message info? Appreciated your comments. Windy ?? ======== 2013-03-25 06:55:07 Original Message? ======== On Sun, Mar 24, 2013 at 9:43 AM, xiaofengcanyuexp at 163.com wrote: Dear freeswitch support I have been studying freeswitch for a few weeks. I will be planning to take the freeswitch as a signal gateway connecting PSTN(ISUP) and SIP. It needs convert the ISUP to SIP based on RFC3372/RFC3204/RFC3398 and vice versa. I notice in the mime_type.cfg supporting applicaiton/ISUP, but I don't find any code in sofia(SIP) module to decode/encode the application/ISUP. My question is: Does freeswitch support to do the signalling gateway connecting ISUP(freeTDM module also has the MTP layer support) and SIP(SIP-T/SIP-I)? If yes, could you let me know how it works? Hello Windy, There is no support in FreeSWITCH for SIP-I or SIP-T ISUP to SIP conversion is supported using Sangoma's SS7 module based on Trillium SS7 stack. Note this is licensed, not open source (this is true for all the MTP layers and ISUP, SCCP etc). Sangoma uses a raw/proprietary mechanism to pass-thru complete IAM messages in a SIP network, it is a crude embedding of the IAM message encoded using base64, within SIP header. We are aware this is crude and by far does not cover all cases, but it was done as quick and dirty way to avoid implementing the whole SIP-I/SIP-T spec and at the same time not miss any IAM information. In all honesty we've had not seen many requests for it so that has kept us from doing the implementation work. Cheers, Moises Silva Manager, Software Engineering msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube = = = = = = = = = = = = = = = = = = = = = = Thanks Windy ?????????????? ?????????????? ??????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/44fe17e3/attachment-0001.html From msc at freeswitch.org Thu May 9 21:22:52 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 May 2013 10:22:52 -0700 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: Since I'm getting ready to submit another order I'll put these on the list. Reminders have a wide range of applications. -MC On Thu, May 9, 2013 at 3:49 AM, Avi Marcus wrote: > Medical isn't the only industry with appointments. > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/a5c7aa85/attachment.html From msc at freeswitch.org Thu May 9 21:30:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 May 2013 10:30:43 -0700 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED In-Reply-To: References: Message-ID: What is your domain name? Go to fs_cli and type: eval ${domain} and see what comes out. then try: sofia_contact */913288@ Last question: are there multiple devices for this person? -Michael On Wed, May 8, 2013 at 9:37 PM, Shiju V.Joseph wrote: > HI , > > sofia_contact can find the user > > freeswitch at internal> sofia_contact 913288 > sofia/internal/sip:913288 at 61.17.226.17:5061 > ;line=f3779717978cb25;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 61.17.226.17:36771 > ;line=5cd95ba4cffdede;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A36771%3E,sofia/internal/sip:913288 at 61.17.226.17:5061 > ;line=6f63c01b0fa6b97;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 124.124.48.35:17394 > ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A17394%3E,sofia/internal/sip:913288 at 124.124.48.35:19746 > ;line=d43c71dfcd69e5b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19746%3E,sofia/internal/sip:913288 at 124.124.48.35:19747 > ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19747%3E > > > but it shows multiple lines , is it normal. > > Thanks > shijujoe > > > On Wed, May 8, 2013 at 3:00 PM, Shiju V.Joseph wrote: > >> Hi All, >> >> I have been experimenting with an odbc based freeswitch cluster in amazon >> ec2 , with opensips doing the load balancing function. >> >> I can make calls to mobile and landlines with out any issues with good >> quality voice , but when i try to call extension to extension freeswith >> shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot >> in lists and wiki and fs jira , tried out different dial strings but with >> out any results.Tried this one >> "dial-string" value="{presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & >> this ${sofia_contact(*/${dialed_user}@${dialed_domain})} >> >> I have copied the siptrace at http://pastebin.com/Qv79tjXK >> >> Appreciate any help in this regard >> >> Thanks >> -- >> shijujoe >> > > > > -- > Shiju V.Joseph > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/3c5a8651/attachment.html From msc at freeswitch.org Thu May 9 21:35:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 May 2013 10:35:45 -0700 Subject: [Freeswitch-users] Call Hangup cause 'None' In-Reply-To: References: Message-ID: Do you have SIP traces of these calls? If not I would get some and see if there's anything explicit in those. It might be helpful in figuring out what is going on. -MC On Thu, May 9, 2013 at 6:32 AM, Rob Moore wrote: > Bump, anyone there ? > > > On Tue, May 7, 2013 at 1:11 PM, Rob Moore wrote: > >> Hi Guys, I was wondering if someone could clear up exactly what the >> hangup case 'none' means? It started appearing after an upgrade to the >> latest version of Freeswitch (from a very very old 1.0.0 version) in place >> of the usual "ALOTTED_TIMEOUT" outcomes we would expect to see from and >> unanswered call. >> >> I would have considered this a simple reclassification of the call result >> however we are still getting a few "ALOTTED_TIMEOUT" mixed in with these >> "none's" which makes me wonder if its actually the symptom of another >> problem I'm not seeing yet. >> >> Any advice on what none means and why it appears in our CDRS would be >> greatly appreciated! >> >> Thanks >> >> Rob >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/a0c8d7b5/attachment.html From anthony.minessale at gmail.com Thu May 9 22:14:15 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 May 2013 13:14:15 -0500 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> <1368013396577-7590468.post@n2.nabble.com> <1368084588897-7590524.post@n2.nabble.com> Message-ID: rtp_timeout_sec in vars.xml will work but it will be global the profile param one will work but only if you update cos I found a bug in it. On Thu, May 9, 2013 at 4:01 AM, Steven Ayre wrote: > No, it'll be within the tag of the sofia . If you > put it in vars.xml it'll be in completely the wrong place and > mod_sofia won't see it... > > > > On 9 May 2013 08:29, mehroz wrote: > > Yes not in X-PRE-PROCESS, but with > > param name="rtp-timeout-sec" value="5" in vars.xml. > > > > I tested the scenario switching off the Wifi of both client.. it worked > fine > > and call was terminated exactly after 5 sec. > > > > But today, i have seen another scenario, i.e, when one of the sip mobile > app > > crashes down, it does not terminates that call. Either RTP is still in > flow > > or something is misunderstood, However i can see the crashed app SIP > device > > gone unregistered on freeswitch after ping option fails. !!!! > > > > > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590524.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://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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/a2dd19d2/attachment-0001.html From rzheng at gmail.com Thu May 9 23:17:38 2013 From: rzheng at gmail.com (Richard Zheng) Date: Thu, 9 May 2013 09:17:38 -1000 Subject: [Freeswitch-users] Freeswitch silent crash In-Reply-To: References: <430001ce41ea$276f6e90$764e4bb0$@ccs.covici.com> <4BF21418-AEAF-4C79-A8C0-28984504B84E@freeswitch.org> <477401ce4499$47347bb0$d59d7310$@gmail.com> Message-ID: Well, my setting was like that already. It still got random crash. What troubles me is not the crash (sXXt happens), it's the lack of core dump. I didn't create the jira because my version is 1.2.3 and obviously I need to try a newer version. But that option is easier said than done... On Wed, May 8, 2013 at 5:39 PM, Gabriel Gunderson wrote: > On Tue, Apr 30, 2013 at 6:57 AM, Antoine Reversat > wrote: > > I changed some odbc settings, I'll let you know how it goes. > > Most of this conversation should have been in Jira, but it wasn't. If > the casual reader comes along, he'll see no resolution to this thread. > So, I quote from the ticket: > > """ > On the mailing list Anthony asked my to make sure I have the right > settings in odbc.ini and odbcinst.ini I was missing those parameters > so I added them : > > UsageCount = 1 > FileUsage = 1 > Threading = 0 > > It hasn't crashed since. > """ > > The take away? Always report bugs in Jira & PostgreSQL FTW... OK, > maybe not the last part. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/4a82b11a/attachment.html From steveayre at gmail.com Thu May 9 23:20:09 2013 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 9 May 2013 20:20:09 +0100 Subject: [Freeswitch-users] Dead channels in freeswitch In-Reply-To: References: <97EB0AF0-86F2-43F4-BFBB-971C59F19912@jerris.com> <1368013396577-7590468.post@n2.nabble.com> <1368084588897-7590524.post@n2.nabble.com> Message-ID: <346F9269-B9C4-4CE3-A7FA-BDE415A2EA51@gmail.com> Ah ok :) hadn't tried it there Steve On 9 May 2013, at 19:14, Anthony Minessale wrote: > rtp_timeout_sec in vars.xml will work but it will be global > the profile param one will work but only if you update cos I found a bug in it. > > > > On Thu, May 9, 2013 at 4:01 AM, Steven Ayre wrote: >> No, it'll be within the tag of the sofia . If you >> put it in vars.xml it'll be in completely the wrong place and >> mod_sofia won't see it... >> >> >> >> On 9 May 2013 08:29, mehroz wrote: >> > Yes not in X-PRE-PROCESS, but with >> > param name="rtp-timeout-sec" value="5" in vars.xml. >> > >> > I tested the scenario switching off the Wifi of both client.. it worked fine >> > and call was terminated exactly after 5 sec. >> > >> > But today, i have seen another scenario, i.e, when one of the sip mobile app >> > crashes down, it does not terminates that call. Either RTP is still in flow >> > or something is misunderstood, However i can see the crashed app SIP device >> > gone unregistered on freeswitch after ping option fails. !!!! >> > >> > >> > >> > -- >> > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dead-channels-in-freeswitch-tp7590259p7590524.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://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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/f9b33a4e/attachment.html From antoine at fibrenoire.ca Thu May 9 23:23:29 2013 From: antoine at fibrenoire.ca (Antoine Reversat) Date: Thu, 9 May 2013 15:23:29 -0400 Subject: [Freeswitch-users] Freeswitch silent crash In-Reply-To: References: <430001ce41ea$276f6e90$764e4bb0$@ccs.covici.com> <4BF21418-AEAF-4C79-A8C0-28984504B84E@freeswitch.org> <477401ce4499$47347bb0$d59d7310$@gmail.com> Message-ID: What fixed the not having a core dump part was adding this option to my sysctl : fs.suid_dumpable = 2 On Thu, May 9, 2013 at 3:17 PM, Richard Zheng wrote: > Well, my setting was like that already. It still got random crash. What > troubles me is not the crash (sXXt happens), it's the lack of core dump. I > didn't create the jira because my version is 1.2.3 and obviously I need to > try a newer version. But that option is easier said than done... > > > On Wed, May 8, 2013 at 5:39 PM, Gabriel Gunderson wrote: > >> On Tue, Apr 30, 2013 at 6:57 AM, Antoine Reversat >> wrote: >> > I changed some odbc settings, I'll let you know how it goes. >> >> Most of this conversation should have been in Jira, but it wasn't. If >> the casual reader comes along, he'll see no resolution to this thread. >> So, I quote from the ticket: >> >> """ >> On the mailing list Anthony asked my to make sure I have the right >> settings in odbc.ini and odbcinst.ini I was missing those parameters >> so I added them : >> >> UsageCount = 1 >> FileUsage = 1 >> Threading = 0 >> >> It hasn't crashed since. >> """ >> >> The take away? Always report bugs in Jira & PostgreSQL FTW... OK, >> maybe not the last part. >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Antoine Reversat Fibrenoire - www.fibrenoire.ca A: 550 ave Beaumont, suite 320, Montreal, Qc, H3N 1V1 T: 514 907-3002 x132 antoine at fibrenoire.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/da8a45fb/attachment-0001.html From bdfoster at davri.com Thu May 9 23:51:08 2013 From: bdfoster at davri.com (Brian Foster) Date: Thu, 9 May 2013 15:51:08 -0400 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: That's probably not the only prompts that could be recorded for reminders, but it's a start. I personally dont have a use for them so I haven't thought long and hard about what prompts are needed. Back to the OP's question, there are many ways to to accomplish what you're trying to do. One thing to take in consideration are the interfaces available on the records system, which could make your life easier. A possible solution might be to use the event socket along with a web service. The latest version of v1.2.stable is recommended right now I believe but you can check http://wiki.freeswitch.org 's main page for the details. I personally recommended Debian 6 for your operating system. A few of the developers recommend it too, but we all have our opinions on what an operating system should look like :) Welcome to FreeSWITCH. You're gonna love it. -BDF On May 9, 2013 1:29 PM, "Michael Collins" wrote: > Since I'm getting ready to submit another order I'll put these on the > list. Reminders have a wide range of applications. > -MC > > > On Thu, May 9, 2013 at 3:49 AM, Avi Marcus wrote: > >> Medical isn't the only industry with appointments. >> >> -Avi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130509/2698b153/attachment.html From schoch+freeswitch.org at xwin32.com Thu May 9 23:59:37 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Thu, 9 May 2013 12:59:37 -0700 Subject: [Freeswitch-users] DTMF Message-ID: One of my users complains of many problems calling automated systems that take DTMF input. In this case it was an IRS payment entry system that would skip or duplicate button presses. Our setup is pretty generic: Polycom SoundPoint IP 320 SIP -> dedicated LAN -> Freeswitch (on CentOS, on XenServer) -> Comcast static IP -> Flowroute.com -> some IRS 800 number. Is this just the way it is, or is there something I can do to make it work better? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/b568a887/attachment.html From msc at freeswitch.org Fri May 10 00:14:00 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 May 2013 13:14:00 -0700 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: Are you sending in-band DTMFs or RFC2833? Or both? Get a pcap of the call, including the RTP, and analyze it in Wireshark. See what you are actually sending. That means looking at the flow (to see if you send RTPEVENTS, i.e. RFC2833 DTMFs) and also listening to the decoded audio with the player. -MC On Thu, May 9, 2013 at 12:59 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > One of my users complains of many problems calling automated systems that > take DTMF input. In this case it was an IRS payment entry system that would > skip or duplicate button presses. > > Our setup is pretty generic: > Polycom SoundPoint IP 320 SIP -> dedicated LAN -> Freeswitch (on CentOS, > on XenServer) -> Comcast static IP -> Flowroute.com -> some IRS 800 number. > > Is this just the way it is, or is there something I can do to make it work > better? > > -- > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/718626cc/attachment.html From luis.daniel.lucio at gmail.com Fri May 10 02:46:20 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 9 May 2013 18:46:20 -0400 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: 2013/5/5 Deon Vermeulen > ASTPP Deon, thank you. Can you talk me about your exerience and how fast they are on fixing bugs. I have a really good or bad luck to find bugs. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/7c5f2233/attachment.html From luis.daniel.lucio at gmail.com Fri May 10 02:54:46 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 9 May 2013 18:54:46 -0400 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: Thank you everyone 2013/5/9 Luis Daniel Lucio Quiroz > > 2013/5/5 Deon Vermeulen > >> ASTPP > > > Deon, thank you. Can you talk me about your exerience and how fast they > are on fixing bugs. I have a really good or bad luck to find bugs. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130509/3ea1c921/attachment.html From mishehu at freeswitch.org Fri May 10 09:16:45 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Fri, 10 May 2013 00:16:45 -0500 Subject: [Freeswitch-users] how Curl post dynamic variables In-Reply-To: <1368002192994-7590463.post@n2.nabble.com> References: <1368002192994-7590463.post@n2.nabble.com> Message-ID: <518C82BD.2060704@freeswitch.org> Why are you exporting new=${res} here? All of these functions appear to operate on the current channel, so you should be able to simply do: Unless I'm missing something.... :-) -Yossi On 05/08/2013 03:36 AM, rajat wrote: > Hi > > I am new to freeswitch and i am trying to integrate one php application with > freeswitch. Purpose is freeswitch will send the user dialed (dtmf numbers) > as input to this php application and in return it will send the response > back. > > Dialplan which I have written for this is > > > > > > > > > > > > > > > > > > Now http://192.168.10.27/index.php this is a url on which i am sending my > requests. Problem here is curl is not taking variable in the post. If > instead of ${new}, i put some random static value like 256, then it is > working perfectly fine but I want my users to input the values. Kindly > suugest some way via which I can post variables in curl url > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/how-Curl-post-dynamic-variables-tp7590463.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://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 From letterstack at gmail.com Fri May 10 09:17:25 2013 From: letterstack at gmail.com (Shiju V.Joseph) Date: Fri, 10 May 2013 10:47:25 +0530 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED In-Reply-To: References: Message-ID: eval and sofia_contact returned the following freeswitch at internal> eval ${domain} ec2-54-242-200-55.compute-1.amazonaws.com freeswitch at internal> freeswitch at internal> sofia_contact */ 913288 at ec2-54-242-200-55.compute-1.amazonaws.com sofia/internal/sip:913288 at 107.20.242.28:38510 ;line=89c9013ed42887b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A107.20.242.28%3A38510%3E freeswitch at internal> I had registered this particular extension from different devices for testing On Thu, May 9, 2013 at 10:07 AM, Shiju V.Joseph wrote: > HI , > > sofia_contact can find the user > > freeswitch at internal> sofia_contact 913288 > sofia/internal/sip:913288 at 61.17.226.17:5061 > ;line=f3779717978cb25;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 61.17.226.17:36771 > ;line=5cd95ba4cffdede;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A36771%3E,sofia/internal/sip:913288 at 61.17.226.17:5061 > ;line=6f63c01b0fa6b97;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A61.17.226.17%3A5061%3E,sofia/internal/sip:913288 at 124.124.48.35:17394 > ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A17394%3E,sofia/internal/sip:913288 at 124.124.48.35:19746 > ;line=d43c71dfcd69e5b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19746%3E,sofia/internal/sip:913288 at 124.124.48.35:19747 > ;line=fa409285164de3c;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A124.124.48.35%3A19747%3E > > > but it shows multiple lines , is it normal. > > Thanks > shijujoe > > > On Wed, May 8, 2013 at 3:00 PM, Shiju V.Joseph wrote: > >> Hi All, >> >> I have been experimenting with an odbc based freeswitch cluster in amazon >> ec2 , with opensips doing the load balancing function. >> >> I can make calls to mobile and landlines with out any issues with good >> quality voice , but when i try to call extension to extension freeswith >> shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot >> in lists and wiki and fs jira , tried out different dial strings but with >> out any results.Tried this one >> "dial-string" value="{presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & >> this ${sofia_contact(*/${dialed_user}@${dialed_domain})} >> >> I have copied the siptrace at http://pastebin.com/Qv79tjXK >> >> Appreciate any help in this regard >> >> Thanks >> -- >> shijujoe >> > > > > -- > Shiju V.Joseph > -- Shiju V.Joseph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/7942ee8d/attachment.html From vermeulen.deon at gmail.com Fri May 10 11:55:09 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Fri, 10 May 2013 08:55:09 +0100 Subject: [Freeswitch-users] Billing software In-Reply-To: References: Message-ID: <518CA7DD.8020802@gmail.com> Hi Luis We just bought Commercial services from ASTPP. When it comes to comparing price, features this is the BEST solution out there at the moment. This is a very active project and you can contact Samir Doshi, Project Maintainer & Developer, directly for more information wrt your technical questions. samir at astpp.org Kind Regards Deon > Luis Daniel Lucio Quiroz > May 9, 2013 11:46 PM > > > Deon, thank you. Can you talk me about your exerience and how fast > they are on fixing bugs. I have a really good or bad luck to find bugs. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > Deon Vermeulen > May 5, 2013 4:52 PM > Check out ASTPP > > > Kind Regards > Deon Vermeulen > > Sent from my iPhone > > Luis Daniel Lucio Quiroz > May 5, 2013 3:36 PM > What other options for FS compatible software other than vBilling do > you recommend me? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/e483a871/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: compose-unknown-contact.jpg Type: image/jpeg Size: 770 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/e483a871/attachment.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: postbox-contact.jpg Type: image/jpeg Size: 1143 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/e483a871/attachment-0001.jpg From steveayre at gmail.com Fri May 10 13:22:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 10 May 2013 10:22:32 +0100 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: > > I personally recommended Debian 6 for your operating system Debian 7 'Wheezy' was released this weekend just gone, and is now the new 'stable' release. Debian 6 is 'Squeeze', which is now 'oldstable'. That said it's far from EOL yet and will continue to receive updates for some time (> 1 year). If you're wanting to roll out a new production system *now* Squeeze *may* still be the best option, just because most people will only have been running Wheezy for a few days so there may be some initial teething problems. That said it's been in release freeze for about 11 months while preparing the release, tracking down and fixing the major bugs so it should be pretty stable. Any new development you're doing that's not for immediate rollout should probably target Wheezy, since you'd have to upgrade to it when Squeeze goes EOL anyway. -Steve On 9 May 2013 20:51, Brian Foster wrote: > That's probably not the only prompts that could be recorded for reminders, > but it's a start. I personally dont have a use for them so I haven't thought > long and hard about what prompts are needed. > > Back to the OP's question, there are many ways to to accomplish what you're > trying to do. One thing to take in consideration are the interfaces > available on the records system, which could make your life easier. A > possible solution might be to use the event socket along with a web service. > The latest version of v1.2.stable is recommended right now I believe but you > can check http://wiki.freeswitch.org 's main page for the details. I > personally recommended Debian 6 for your operating system. A few of the > developers recommend it too, but we all have our opinions on what an > operating system should look like :) > > Welcome to FreeSWITCH. You're gonna love it. > > -BDF > > On May 9, 2013 1:29 PM, "Michael Collins" wrote: >> >> Since I'm getting ready to submit another order I'll put these on the >> list. Reminders have a wide range of applications. >> -MC >> >> >> On Thu, May 9, 2013 at 3:49 AM, Avi Marcus wrote: >>> >>> Medical isn't the only industry with appointments. >>> >>> -Avi >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/f8077d9f/attachment-0001.html From jeff at askcornerstone.net Fri May 10 13:02:53 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Fri, 10 May 2013 09:02:53 +0000 Subject: [Freeswitch-users] Google Voice Question Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CB7EB@mail2.firstdataworks.net> Setting up to use my Google Voice number (outbound only for now) according to this thread: http://www.broadbandreports.com/forum/r27205004-FreeSWITCH-Google-Voice-Setup I found it more thorough than the Wiki tutorial (though there's the problem of case sensitivity in the "GTalk" profile name). I've gotten as far as getting outbound calls to ring... BUT they don't ring to the phone I'm dialing, so not sure why... wonder where they're ringing to! I just let it ring 'til Sofia kills it. Here's the log: http://pastebin.freeswitch.org/20902 (take 1.2.3.4 as my ext-ip). What I wonder about is the series of messages that say "Accepted 0 of 0 ... candidates." Is this where it's falling apart? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/e6d7ffd8/attachment.html From alex at digitalmail.com Fri May 10 16:03:11 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 10 May 2013 13:03:11 +0100 Subject: [Freeswitch-users] set_tts_params Message-ID: <518CE1FF.1050205@digitalmail.com> Can anyone tell me in which version of Freeswitch set_tts_params function was introduced? Although set_tts_parms is now deprecated, I seem to have some 1.2 installations that don't have it! From mburakbor at gmail.com Fri May 10 15:30:05 2013 From: mburakbor at gmail.com (=?ISO-8859-9?Q?Burak_BorYaz=FDl=FDm?=) Date: Fri, 10 May 2013 14:30:05 +0300 Subject: [Freeswitch-users] Freeswitch User Adding Message-ID: Hello, I have problems with adding new user to freeswitch. When trying to add user with user id has different number of digits than 4, it can register but it cant call or cant be called. I changed local extension regular expression in dialpan default.xml but the changes only work with four digits users(user ids or dial number) .So I want dial a number that has five or more digits. What other configurations I must change. Thank you... Burak, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/42728da1/attachment.html From sertys at gmail.com Fri May 10 18:59:42 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 10 May 2013 16:59:42 +0200 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: Well, make sure your calls actually hit the default context not for example the public. And how exactly did you change the regex? Also did you do reloadxml after modifying .xml files? On May 10, 2013 5:36 PM, "Burak BorYaz?l?m" wrote: > Hello, > > I have problems with adding new user to freeswitch. When trying to add > user with user id has different number of digits than 4, it can register > but it cant call or cant be called. I changed local extension regular > expression in dialpan default.xml but the changes only work with four > digits users(user ids or dial number) .So I want dial a number that has > five or more digits. What other configurations I must change. > > Thank you... > Burak, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/a1d2f30a/attachment.html From mburakbor at gmail.com Fri May 10 19:11:57 2013 From: mburakbor at gmail.com (=?ISO-8859-9?Q?Burak_BorYaz=FDl=FDm?=) Date: Fri, 10 May 2013 18:11:57 +0300 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: Yes I used reloadxml after changing. Also the changing the regex works perfect with four digits numbers. It is tested well. But can you explain more your first statement please? 2013/5/10 Daniel Ivanov > Well, make sure your calls actually hit the default context not for > example the public. And how exactly did you change the regex? Also did you > do reloadxml after modifying .xml files? > On May 10, 2013 5:36 PM, "Burak BorYaz?l?m" wrote: > >> Hello, >> >> I have problems with adding new user to freeswitch. When trying to add >> user with user id has different number of digits than 4, it can register >> but it cant call or cant be called. I changed local extension regular >> expression in dialpan default.xml but the changes only work with four >> digits users(user ids or dial number) .So I want dial a number that has >> five or more digits. What other configurations I must change. >> >> Thank you... >> Burak, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/f0cf9a5d/attachment.html From sdevoy at bizfocused.com Fri May 10 20:00:54 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 10 May 2013 12:00:54 -0400 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: <101401ce4d97$8d6db220$a8491660$@bizfocused.com> Two things come to mind Burak. Daniel is correct, double check your directory xml to be sure the context="??????" is the same so you are using the same FreeSwitch dialplan. Second, I had a similar problem because the phone's dialplan (not Freeswitch, but in the phone's configuration) has to allow 5 digit dialing. What type of phone (or ata) are you using? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Burak BorYazilim Sent: Friday, May 10, 2013 11:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch User Adding Yes I used reloadxml after changing. Also the changing the regex works perfect with four digits numbers. It is tested well. But can you explain more your first statement please? 2013/5/10 Daniel Ivanov Well, make sure your calls actually hit the default context not for example the public. And how exactly did you change the regex? Also did you do reloadxml after modifying .xml files? On May 10, 2013 5:36 PM, "Burak BorYaz?l?m" wrote: Hello, I have problems with adding new user to freeswitch. When trying to add user with user id has different number of digits than 4, it can register but it cant call or cant be called. I changed local extension regular expression in dialpan default.xml but the changes only work with four digits users(user ids or dial number) .So I want dial a number that has five or more digits. What other configurations I must change. Thank you... Burak, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130510/0a6ea0f2/attachment-0001.html From sdevoy at bizfocused.com Fri May 10 20:11:04 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 10 May 2013 12:11:04 -0400 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: <102e01ce4d98$f93e4ae0$ebbae0a0$@bizfocused.com> Can you post the log file to pastebin.freeswitch.org where you call to and from these extensions? Let us know the link to your pastebin. Also post the result of "Show registrations". Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Burak BorYazilim Sent: Friday, May 10, 2013 11:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch User Adding Yes I used reloadxml after changing. Also the changing the regex works perfect with four digits numbers. It is tested well. But can you explain more your first statement please? 2013/5/10 Daniel Ivanov Well, make sure your calls actually hit the default context not for example the public. And how exactly did you change the regex? Also did you do reloadxml after modifying .xml files? On May 10, 2013 5:36 PM, "Burak BorYaz?l?m" wrote: Hello, I have problems with adding new user to freeswitch. When trying to add user with user id has different number of digits than 4, it can register but it cant call or cant be called. I changed local extension regular expression in dialpan default.xml but the changes only work with four digits users(user ids or dial number) .So I want dial a number that has five or more digits. What other configurations I must change. Thank you... Burak, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130510/0749c2e8/attachment.html From steveayre at gmail.com Fri May 10 20:29:51 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 10 May 2013 17:29:51 +0100 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: Are you using the default dialplan? Chances are your destination_number condition's regex for the extension that calls users is limited to 4 digits. Eg \d\d\d\d \d{4} -Steve On 10 May 2013 12:30, Burak BorYaz?l?m wrote: > Hello, > > I have problems with adding new user to freeswitch. When trying to add > user with user id has different number of digits than 4, it can register > but it cant call or cant be called. I changed local extension regular > expression in dialpan default.xml but the changes only work with four > digits users(user ids or dial number) .So I want dial a number that has > five or more digits. What other configurations I must change. > > Thank you... > Burak, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/9349e7af/attachment.html From chris.hall at vividapps.co.uk Fri May 10 20:03:37 2013 From: chris.hall at vividapps.co.uk (Christopher Hall) Date: Fri, 10 May 2013 17:03:37 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE Message-ID: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> Hi, I'm new to FreeSWITCH lists and I'm not sure if this is the correct list for my post, any help will be much appreciated. I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don't see any kind of response. Help! Thanks Chris. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/7cb7e1ab/attachment.html From alex at digitalmail.com Fri May 10 21:16:02 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 10 May 2013 18:16:02 +0100 Subject: [Freeswitch-users] Node.JS ESL libraries Message-ID: <518D2B52.1090304@digitalmail.com> Just beginning to get a bit more advanced with our use of Freeswitch, and we'd like to start integrating it's internal data and status with the outside world a bit more. The idea was to listen in on the event socket and use that information to update an external data model. Looking at the wiki(http://wiki.freeswitch.org/wiki/Event_Socket#Javascript_.2F_Node.js_library), there seem to be 3 or 4 examples of how this is done, but I wondered if there's stuff that's not there or if one of these methods is proving more popular than the others. Any advice on where we should look first would be appreciated, cheers! From dgarcia at anew.com.ve Fri May 10 22:49:33 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 10 May 2013 14:19:33 -0430 Subject: [Freeswitch-users] Advice using mod_curl Message-ID: <518D413D.301@anew.com.ve> Hi, I have lua script. the script is similar to the one in http://wiki.freeswitch.org/wiki/Mod_curl (lua usage) The script access a web service and get some data. The script work fine for my needs. However, sometimes when the script access the web service take several seconds to get the data, and the caller ear silence. Is it possible to play a sound file as long the script try to get the data from the web service? -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/75bd3ca4/attachment-0001.html From steveayre at gmail.com Fri May 10 23:01:18 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 10 May 2013 20:01:18 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> Message-ID: <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> Try enabling the sip trace when it happens to see if FS receives the packets: sofia global siptrace on Steve On 10 May 2013, at 17:03, "Christopher Hall" wrote: > Hi, > > I?m new to FreeSWITCH lists and I?m not sure if this is the correct list for my post, any help will be much appreciated. > > I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. > > I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. > > However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. > > When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don?t see any kind of response. > > Help! > > Thanks > > Chris. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/a23065f9/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri May 10 23:13:44 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 10 May 2013 20:13:44 +0100 Subject: [Freeswitch-users] Advice using mod_curl In-Reply-To: <518D413D.301@anew.com.ve> References: <518D413D.301@anew.com.ve> Message-ID: Hello, Can I ask why your web service is taking several seconds to respond? Could you not speed this up? As far as I'm aware, telling mod_curl to play music when a request is being executed is really not the right way to look at it. If you need to retrieve data from an external service that will take several seconds, you would place the call into hold or into a parking queue, execute the request, then come back.. however you'd do this within the mod_curl response, rather than it happening on a mod_curl call itself. For example, you could get the mod_curl to response immediately, then use a mixture of httpapi/includes/parking to achieve your goal.. Perhaps someone else might be able to offer a cleaner solution, but that's the best I can think of at the moment. Hope this helps! Cal On Fri, May 10, 2013 at 7:49 PM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > Hi, > > I have lua script. the script is similar to the one in > http://wiki.freeswitch.org/wiki/Mod_curl (lua usage) > > The script access a web service and get some data. The script work fine > for my needs. However, sometimes when the script access the web service > take several seconds to get the data, and the caller ear silence. Is it > possible to play a sound file as long the script try to get the data from > the web service? > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/cc1928f3/attachment.html From sdevoy at bizfocused.com Sat May 11 00:01:14 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 10 May 2013 16:01:14 -0400 Subject: [Freeswitch-users] Caller ID - a moving target Message-ID: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> Hi, When someone dials my extension from an internal extension, my phone displays the CID Number. When I answer, the display switches to CID Name. Is that a "feature" of Cisco SPA504G phones or FS? Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/0fcd80d4/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/0fcd80d4/attachment.gif From jmesquita at freeswitch.org Sat May 11 00:29:04 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 10 May 2013 17:29:04 -0300 Subject: [Freeswitch-users] Caller ID - a moving target In-Reply-To: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> References: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> Message-ID: ignore_display_updates=true will disable this behavior. That's a FS thing and it is supposed to do that as SIP supports cid updates. Jo?o Mesquita FreeSWITCH? Solutions On Fri, May 10, 2013 at 5:01 PM, Sean Devoy wrote: > Hi,**** > > ** ** > > When someone dials my extension from an internal extension, my phone > displays the CID Number. When I answer, the display switches to CID Name. > **** > > ** ** > > Is that a ?feature? of Cisco SPA504G phones or FS?**** > > ** ** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130510/045eecad/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/045eecad/attachment-0001.gif From cal.leeming at simplicitymedialtd.co.uk Sat May 11 00:39:51 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 10 May 2013 21:39:51 +0100 Subject: [Freeswitch-users] Analysis of Cisco 7940, SIP ALG and NAT traversal problems Message-ID: Hello all, I've seen the matter of NAT traversal come up several times, but very few people go into detail about some of the specific problems and how/why they happen. So today I spent a few hours doing a write up of an on-going problem we've had with the Cisco 7940 phones, and a variety of routers. This article goes into quite a bit of detail, giving a breakdown of some of the specific problems we were having, and some of the ways you can resolve it. Although not strictly FreeSWITCH related (despite the FS reference), I'm hoping that this write up will help others understand some of the NAT traversal problems; http://blog.simplicitymedialtd.co.uk/476/analysis-of-cisco-7940-sip-alg-and-nat-traversal-problems Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/24a74113/attachment.html From msc at freeswitch.org Sat May 11 02:02:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 May 2013 15:02:23 -0700 Subject: [Freeswitch-users] Node.JS ESL libraries In-Reply-To: <518D2B52.1090304@digitalmail.com> References: <518D2B52.1090304@digitalmail.com> Message-ID: I don't have any specific examples in mind but I do recommend you check out the freeswitch-contrib repo. People have put a lot of interesting stuff in there. I know that intralaman has some ESL stuff in there that you might use as a reference. -MC On Fri, May 10, 2013 at 10:16 AM, Alex Lake wrote: > Just beginning to get a bit more advanced with our use of Freeswitch, > and we'd like to start integrating it's internal data and status with > the outside world a bit more. > > The idea was to listen in on the event socket and use that information > to update an external data model. > > Looking at the > wiki( > http://wiki.freeswitch.org/wiki/Event_Socket#Javascript_.2F_Node.js_library > ), > there seem to be 3 or 4 examples of how this is done, but I wondered if > there's stuff that's not there or if one of these methods is proving > more popular than the others. > > Any advice on where we should look first would be appreciated, cheers! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/f31742f6/attachment.html From msc at freeswitch.org Sat May 11 02:15:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 May 2013 15:15:09 -0700 Subject: [Freeswitch-users] looking for a way to do appointment reminders In-Reply-To: References: <518AEC0B.8070708@sbcglobal.net> Message-ID: On Fri, May 10, 2013 at 2:22 AM, Steven Ayre wrote: > I personally recommended Debian 6 for your operating system > > > Debian 7 'Wheezy' was released this weekend just gone, and is now the new > 'stable' release. > > Debian 6 is 'Squeeze', which is now 'oldstable'. That said it's far from > EOL yet and will continue to receive updates for some time (> 1 year). > > If you're wanting to roll out a new production system *now* Squeeze *may* > still be the best option, just because most people will only have been > running Wheezy for a few days so there may be some initial teething > problems. That said it's been in release freeze for about 11 months while > preparing the release, tracking down and fixing the major bugs so it should > be pretty stable. > > Any new development you're doing that's not for immediate rollout should > probably target Wheezy, since you'd have to upgrade to it when Squeeze goes > EOL anyway. > > -Steve > Steve, Thanks for the heads up on Deb7. One of these days I will get around to installing it and trying it out. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/ce943354/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat May 11 03:07:36 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 11 May 2013 00:07:36 +0100 Subject: [Freeswitch-users] 5-for-1 review request for FS-3899 (mod_xml_curl error detection) Message-ID: Hello, In line with the 5-for-1 approach mentioned a few months ago, I have triaged FS-3964, FS-4211, FS-4211, FS-4891 and FS-5154. In return, could a core dev please take a moment to look at this ticket: http://jira.freeswitch.org/browse/FS-3899 This started off as a discussion about making FS wait until a remote web server is available before booting up, however I've added some extra comments on a different approach that could be used. My suggestion is to add additional functionality to mod_xml_curl that allows you to define an on_failure event, which then executes any script of your choice.. you can then implement your own logic in the script. This would allow you to detect mod_xml_curl problems, and stop your FS instance from being in a 'semi loaded' state in the event of a temporary web server problem. Any thoughts/feedback would be much appreciated. Many thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130511/cea5bf9f/attachment.html From sdevoy at bizfocused.com Sat May 11 05:31:50 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 10 May 2013 21:31:50 -0400 Subject: [Freeswitch-users] Caller ID - a moving target In-Reply-To: References: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> Message-ID: <138701ce4de7$4fad61b0$ef082510$@bizfocused.com> Thank you Jo?o, but you don?t get off that easy! Can I get CID Name and Number displayed in either/both cases? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Friday, May 10, 2013 4:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Caller ID - a moving target ignore_display_updates=true will disable this behavior. That's a FS thing and it is supposed to do that as SIP supports cid updates. Jo?o Mesquita FreeSWITCH? Solutions On Fri, May 10, 2013 at 5:01 PM, Sean Devoy wrote: Hi, When someone dials my extension from an internal extension, my phone displays the CID Number. When I answer, the display switches to CID Name. Is that a ?feature? of Cisco SPA504G phones or FS? Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130510/467d16ae/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130510/467d16ae/attachment-0001.gif From jeff at jefflenk.com Sat May 11 06:17:11 2013 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 10 May 2013 19:17:11 -0700 (PDT) Subject: [Freeswitch-users] Google Voice Question In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CB7EB@mail2.firstdataworks.net> References: <8A9716A5B256904FB1F07C050F9CCCCB020CB7EB@mail2.firstdataworks.net> Message-ID: <1368238631299-7590573.post@n2.nabble.com> remove the H264 codec from the list in dingaling.conf.xml - video doesn't work with google voice. Also make sure you are using the latest code from git for stable or master because there was a commit today to fix a problem. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-Question-tp7590553p7590573.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at askcornerstone.net Sat May 11 08:04:24 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Sat, 11 May 2013 04:04:24 +0000 Subject: [Freeswitch-users] Google Voice Question In-Reply-To: <1368238631299-7590573.post@n2.nabble.com> References: <8A9716A5B256904FB1F07C050F9CCCCB020CB7EB@mail2.firstdataworks.net> <1368238631299-7590573.post@n2.nabble.com> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CBED3@mail2.firstdataworks.net> Cool. I'll try it out. Thank you! Jeff Bernhardt Systems Administrator Cornerstone Consulting 808.440.2900 -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Friday, May 10, 2013 4:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Google Voice Question remove the H264 codec from the list in dingaling.conf.xml - video doesn't work with google voice. Also make sure you are using the latest code from git for stable or master because there was a commit today to fix a problem. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-Question-tp7590553p7590573.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://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 From yudha2008 at gmail.com Sat May 11 09:43:50 2013 From: yudha2008 at gmail.com (baskar) Date: Fri, 10 May 2013 22:43:50 -0700 (PDT) Subject: [Freeswitch-users] NO_ROUTE_DESTINATION In-Reply-To: References: <1368008894683-7590466.post@n2.nabble.com> Message-ID: <1368251030474-7590575.post@n2.nabble.com> Hi Steve, Thanks for the reply. When i try to use only the gateway it pass through external profile why it is happning. SIP trunk has username and password with hard cable connected from service provider. SIP trunk configured in freeswitch server and pass through service provider through SIP trunk. sofia/internal/407 at 10.15.1.41 bridge(sofia/gateway/xxxx/0018444485452) 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f] Can any one guide me procedure to calls passing through the SIP server. Thanks in advance. Regards, N.Baskar -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.html Sent from the freeswitch-users mailing list archive at Nabble.com. From chris.hall at vividapps.co.uk Sat May 11 12:24:55 2013 From: chris.hall at vividapps.co.uk (Christopher Hall) Date: Sat, 11 May 2013 09:24:55 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> Message-ID: <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> Steve, Thanks for your response, I have tried with global siptrace turned on and I see absolutely nothing when I make the call, the SIP INVITE messages are sent to the FreeSWITCH host machine as seen using WireShark running on that host. Chris. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre Sent: 10 May 2013 20:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE Try enabling the sip trace when it happens to see if FS receives the packets: sofia global siptrace on Steve On 10 May 2013, at 17:03, "Christopher Hall" wrote: Hi, I?m new to FreeSWITCH lists and I?m not sure if this is the correct list for my post, any help will be much appreciated. I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don?t see any kind of response. Help! Thanks Chris. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130511/e24a2385/attachment.html From jeff at askcornerstone.net Sat May 11 12:50:03 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Sat, 11 May 2013 08:50:03 +0000 Subject: [Freeswitch-users] Google Voice Question In-Reply-To: <1368238631299-7590573.post@n2.nabble.com> References: <8A9716A5B256904FB1F07C050F9CCCCB020CB7EB@mail2.firstdataworks.net>, <1368238631299-7590573.post@n2.nabble.com> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CBF45@mail2.firstdataworks.net> Don't know if it was disabling H264 or the update, but it works now! Now to try getting inbound to work... Thanks again. ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Jeff Lenk [jeff at jefflenk.com] Sent: Friday, May 10, 2013 4:17 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Google Voice Question remove the H264 codec from the list in dingaling.conf.xml - video doesn't work with google voice. Also make sure you are using the latest code from git for stable or master because there was a commit today to fix a problem. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Google-Voice-Question-tp7590553p7590573.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://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 From xiaofengcanyuexp at 163.com Sat May 11 14:09:06 2013 From: xiaofengcanyuexp at 163.com (=?utf-8?B?eGlhb2ZlbmdjYW55dWV4cEAxNjMuY29t?=) Date: Sat, 11 May 2013 18:09:06 +0800 Subject: [Freeswitch-users] =?utf-8?q?_How_to_integrate_the_private_code_t?= =?utf-8?q?o_sofia_and_freetdm=3F?= References: <201303242132282818393@163.com>, <201303242142599068438@163.com>, , <201305092236322508083@163.com> Message-ID: <201305111809048430044@163.com> Dear freeswitch support?? As you suggested, I have completed the ISUP<->SIP-T conversion work. While there are some problems to integrate my code to freeswitch. 1. I got one sangoma license and can bridge the freetmd to sofia now. But how can I intercept the ISUP mesasge(from the freetdm) and run to my application and then to Sofia module? Is there any external control like ESL to do the work? The diagram is like: SIP-T ISUP SofiaModule<-------------My Conversion---------------------->freetdm 2. I have been working with ESL "SEND" command to control the Sofia module, while it needs to divide the conversion SIP-T message. Is there anyway to put the one whole SIP-T message to Sofia module to handle? like one external SIP-T message to freeswitch? And vice versa, Is there good way to receive the whole SIP message from sofia module instead of using ESL to detect the SIP message info? Appreciated your comments. Windy ?? ======== 2013-03-25 06:55:07 Original Message? ======== On Sun, Mar 24, 2013 at 9:43 AM, xiaofengcanyuexp at 163.com wrote: Dear freeswitch support I have been studying freeswitch for a few weeks. I will be planning to take the freeswitch as a signal gateway connecting PSTN(ISUP) and SIP. It needs convert the ISUP to SIP based on RFC3372/RFC3204/RFC3398 and vice versa. I notice in the mime_type.cfg supporting applicaiton/ISUP, but I don't find any code in sofia(SIP) module to decode/encode the application/ISUP. My question is: Does freeswitch support to do the signalling gateway connecting ISUP(freeTDM module also has the MTP layer support) and SIP(SIP-T/SIP-I)? If yes, could you let me know how it works? Hello Windy, There is no support in FreeSWITCH for SIP-I or SIP-T ISUP to SIP conversion is supported using Sangoma's SS7 module based on Trillium SS7 stack. Note this is licensed, not open source (this is true for all the MTP layers and ISUP, SCCP etc). Sangoma uses a raw/proprietary mechanism to pass-thru complete IAM messages in a SIP network, it is a crude embedding of the IAM message encoded using base64, within SIP header. We are aware this is crude and by far does not cover all cases, but it was done as quick and dirty way to avoid implementing the whole SIP-I/SIP-T spec and at the same time not miss any IAM information. In all honesty we've had not seen many requests for it so that has kept us from doing the implementation work. Cheers, Moises Silva Manager, Software Engineering msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube = = = = = = = = = = = = = = = = = = = = = = Thanks Windy ?????????????? ?????????????? ??????????????? = = = = = = = = = = = = = = = = = = = = = = Thanks Windy ?????????????? ?????????????? ??????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130511/9c008f29/attachment-0001.html From emamirazavi at gmail.com Sat May 11 15:23:09 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Sat, 11 May 2013 15:53:09 +0430 Subject: [Freeswitch-users] An strange bug on starting up FreeSwitch, any solution? Message-ID: after 10 seconds FS goes down! below is my syslogs: [root at Ctel freeswitch]# tail -f /var/log/messages May 11 18:55:00 aratel kernel: freeswitch[11261]: segfault at 1 ip 00828282 sp b70539e8 error 4 in libc-2.12.so[6f7000+189000] May 11 18:55:00 aratel abrt[11262]: saved core dump of pid 11193 (/usr/local/freeswitch/bin/freeswitch) to /var/spool/abrt/ccpp-2013-05-11-18:55:00-11193.new/coredump (8544256 bytes) May 11 18:55:00 aratel abrtd: Directory 'ccpp-2013-05-11-18:55:00-11193' creation detected May 11 18:55:00 aratel abrtd: Executable '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package May 11 18:55:00 aratel abrtd: Corrupted or bad dump /var/spool/abrt/ccpp-2013-05-11-18:55:00-11193 (res:2), deleting May 11 19:17:23 aratel kernel: freeswitch[11342]: segfault at 1 ip 00828282 sp b6e859e8 error 4 in libc-2.12.so[6f7000+189000] May 11 19:17:23 aratel abrt[11343]: saved core dump of pid 11330 (/usr/local/freeswitch/bin/freeswitch) to /var/spool/abrt/ccpp-2013-05-11-19:17:23-11330.new/coredump (8540160 bytes) May 11 19:17:23 aratel abrtd: Directory 'ccpp-2013-05-11-19:17:23-11330' creation detected May 11 19:17:23 aratel abrtd: Executable '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package May 11 19:17:23 aratel abrtd: Corrupted or bad dump /var/spool/abrt/ccpp-2013-05-11-19:17:23-11330 (res:2), deleting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130511/023341db/attachment.html From mburakbor at gmail.com Sat May 11 15:28:00 2013 From: mburakbor at gmail.com (=?ISO-8859-9?Q?Burak_BorYaz=FDl=FDm?=) Date: Sat, 11 May 2013 14:28:00 +0300 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: First off all thanks for your kind helps. Let me tell you the situation more clearly. Firstly I used Android phones as sip clients with the open source sip client program, CSipSimple. My server computer has Ubuntu 12.04 LTS with one static ip address. freeswitch version output: FreeSWITCH Version 1.3.13b+git~20130205T003128Z~70a9560306 (git 70a9560 2013-02-05 00:31:28Z) We were using the system with 4 digits number sip account names without any problem. Also the account name will be used as dial number. An example of the user xml file is below. I changed the regex in dialpan/default.xml. The changings were perfectly succesfull with, again, 4 digits number. But when trying to include any other number of digits (3, 5 and 6 were tested), it doesn't work(When I change the regex to accept only 5 digits numbers, 4 digits ones didnt work as expected). I really could not understand why it is happening. Why there was no problem with 4 digits number and why the exact same system does not work with this basic change. To explain the errors more, I want to talk about my tests. Firstly I used 4 digits number user. This test repeated with tls and srtp. These two test were succesfull. And the secand test is same with first test but with 5 digits numbers without tls and srtp. Registration was succesfull but cant call. (or the call could not be forwarded) Of course I changed the regex and execute reloadxml in this test. GSM (8kHz) and G722(16kHz) codecs were used. As you see below, CODEC NEGOTIATION ERROR occured. After getting this error I changed the codes as SILK(16kHz). With this change I made the third test. Third test result is also below. show registretions output: reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata 60022,my.server.ip.address,oOnxiVJFAgQWLABrD01UsYTVOY3TVSlx,sofia/internal/sip:60022 at 141.196.174.60:52245 ;ob,1368271435,141.196.174.60,52245,udp,server, 60021,my.server.ip.address,5r4MTDiZPhs5qzdin9A3hEUh1zZsdqqk,sofia/internal/sip:60021 at 141.196.174.60:57938 ;ob,1368271446,141.196.174.60,57938,udp,server, @Sean, sorry but I cant post my whole log file because my server has got a network attack so I am quite busy with this attacker. And also I dont think CSipSimple has a problem with 5 digits because this system (with 5 digits) was succesfull with Kamailio and CSipSimple. If I could not be clear please let me know. Thanks... user xml example: This was exact same with the default users in freeswitch. Only the number changed. The second test error: 2013-05-11 12:13:56.800443 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 12:13:56.900444 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:1683 Register: From: [60021 at my.server.ip.address] Contact: ["user" ] Expires: [900] 2013-05-11 12:14:06.820443 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:06.900444 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:1683 Register: From: [60022 at my.server.ip.address] Contact: ["user" ] Expires: [900] 2013-05-11 12:14:14.640437 [NOTICE] switch_channel.c:968 New Channel sofia/internal/60021 at my.server.ip.address[26730cd2-ba1b-11e2-acc5-bda7cbfd9554] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_NEW 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/60021 at my.server.ip.address) State NEW 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 12:14:14.660438 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:14.660438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:1719 detaching session 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 2013-05-11 12:14:14.780439 [DEBUG] sofia.c:1811 Re-attaching to session 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel sofia/internal/60021 at my.server.ip.address entering state [received][100] 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5802 (sofia/internal/60021 at my.server.ip.address) State Change CS_NEW -> CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60021 at my.server.ip.address) State INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 sofia/internal/60021 at my.server.ip.address SOFIA INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 (sofia/internal/60021 at my.server.ip.address) State Change CS_INIT -> CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60021 at my.server.ip.address) State INIT going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 (sofia/internal/60021 at my.server.ip.address) Callstate Change DOWN -> RINGING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60021 at my.server.ip.address) State ROUTING 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 sofia/internal/60021 at my.server.ip.address SOFIA ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:117 sofia/internal/60021 at my.server.ip.address Standard ROUTING 2013-05-11 12:14:14.800439 [INFO] mod_dialplan_xml.c:557 Processing 60021 <60021>->60022 in context default Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->unloop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->tod_example] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->holiday_example] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->global-intercept] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global-intercept] destination_number(60022) =~ /^886$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->group-intercept] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [group-intercept] destination_number(60022) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [intercept-ext] destination_number(60022) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->redial] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [redial] destination_number(60022) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->global] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Action set(sip_secure_media=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action export(sip_secure_media=true) Dialplan: sofia/internal/60021 at my.server.ip.address Absolute Condition [global] Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [snom-demo-2] destination_number(60022) =~ /^9001$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [snom-demo-1] destination_number(60022) =~ /^9000$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60022) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60022) =~ /^779$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call_return] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call_return] destination_number(60022) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->del-group] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [del-group] destination_number(60022) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->add-group] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [add-group] destination_number(60022) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call-group-simo] destination_number(60022) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call-group-order] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call-group-order] destination_number(60022) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [extension-intercom] destination_number(60022) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [Local_Extension] destination_number(60022) =~ /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Action export(dialed_extension=60022) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(ringback=${us-ring}) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(call_timeout=30) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(hangup_after_bridge=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(continue_on_fail=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/60021 at my.server.ip.address Action answer() Dialplan: sofia/internal/60021 at my.server.ip.address Action sleep(1000) Dialplan: sofia/internal/60021 at my.server.ip.address Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/60021 at my.server.ip.address) State Change CS_ROUTING -> CS_EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60021 at my.server.ip.address) State ROUTING going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60021 at my.server.ip.address) State EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:242 sofia/internal/60021 at my.server.ip.address SOFIA EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:209 sofia/internal/60021 at my.server.ip.address Standard EXECUTE EXECUTE sofia/internal/60021 at my.server.ip.addressset(sip_secure_media=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [sip_secure_media]=[true] EXECUTE sofia/internal/60021 at my.server.ip.addressexport(sip_secure_media=true) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_secure_media]=[true] EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-spymap/60021/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/60021/60022) EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/global/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address export(RFC2822_DATE=Sat, 11 May 2013 12:14:14 +0300) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] EXECUTE sofia/internal/60021 at my.server.ip.addressexport(dialed_extension=60022) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [dialed_extension]=[60022] EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(1 b s execute_extension::dx XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/60021.2013-05-11-12-14-14.wav) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/60021.2013-05-11-12-14-14.wav EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(3 b s execute_extension::cf XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(4 b s execute_extension::att_xfer XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/60021 at my.server.ip.addressset(ringback=%(2000,4000,440,480)) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/60021 at my.server.ip.addressset(transfer_ringback=local_stream://moh) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/60021 at my.server.ip.address set(call_timeout=30) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [call_timeout]=[30] EXECUTE sofia/internal/60021 at my.server.ip.addressset(hangup_after_bridge=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/60021 at my.server.ip.addressset(continue_on_fail=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [continue_on_fail]=[true] EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-call_return/60022/60021) EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/60022/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.addressset(called_party_callgroup=techsupport) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/techsupport/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/global/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/techsupport/26730cd2-ba1b-11e2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.addressbridge(user/60022 at my.server.ip.address) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60022] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60022] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 12:14:14.800439 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[my.server.ip.address] 2013-05-11 12:14:14.800439 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[60022 at my.server.ip.address] 2013-05-11 12:14:14.800439 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:60022 at 141.196.174.60:33822[268c95ee-ba1b-11e2-ace5-bda7cbfd9554] 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:4961 (sofia/internal/ sip:60022 at 141.196.174.60:33822) State Change CS_NEW -> CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:5031 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:5034 [zrtp_passthru] Setting b-leg absolute_codec_string='PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b ,GSM at 8000h@20i at 13200b' 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60022 at 141.196.174.60:33822) State INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 sofia/internal/ sip:60022 at 141.196.174.60:33822 SOFIA INIT 2013-05-11 12:14:14.800439 [DEBUG] sofia_glue.c:3157 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:jy7Mnu44PUrnS4nFSUGkaIsFNftmZRTnE61m4sui] 2013-05-11 12:14:14.800439 [DEBUG] sofia_glue.c:2649 Local SDP: v=0 o=FreeSWITCH 1368242802 1368242803 IN IP4 my.server.ip.address s=FreeSWITCH c=IN IP4 my.server.ip.address t=0 0 m=audio 20852 RTP/SAVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:jy7Mnu44PUrnS4nFSUGkaIsFNftmZRTnE61m4sui a=ptime:20 a=sendrecv m=audio 20852 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 (sofia/internal/ sip:60022 at 141.196.174.60:33822) State Change CS_INIT -> CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60022 at 141.196.174.60:33822) State INIT going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 (sofia/internal/ sip:60022 at 141.196.174.60:33822) Callstate Change DOWN -> RINGING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60022 at 141.196.174.60:33822) State ROUTING 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 sofia/internal/ sip:60022 at 141.196.174.60:33822 SOFIA ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60022 at 141.196.174.60:33822) State ROUTING going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60022 at 141.196.174.60:33822) State CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60022 at 141.196.174.60:33822) State CONSUME_MEDIA going to sleep 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel sofia/internal/ sip:60022 at 141.196.174.60:33822 entering state [calling][0] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] sofia.c:5578 Channel sofia/internal/ sip:60022 at 141.196.174.60:33822 entering state [proceeding][180] 2013-05-11 12:14:14.980444 [NOTICE] sofia.c:5670 Ring-Ready sofia/internal/ sip:60022 at 141.196.174.60:33822! 2013-05-11 12:14:14.980444 [INFO] switch_ivr_originate.c:1185 Sending early media 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, refer to rfc3711 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> HANGUP 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/60021 at my.server.ip.address [KILL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:1186 sofia/internal/60021 at my.server.ip.address Media Establishment Failed. 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 (sofia/internal/ sip:60022 at 141.196.174.60:33822) Callstate Change RINGING -> HANGUP 2013-05-11 12:14:14.980444 [NOTICE] switch_ivr_originate.c:3620 Hangup sofia/internal/sip:60022 at 141.196.174.60:33822 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_HANGUP 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [KILL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60022 at 141.196.174.60:33822) State HANGUP 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ sip:60022 at 141.196.174.60:33822 hanging up, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [INFO] mod_dptools.c:3060 Originate Failed. Cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:562 Sending CANCEL to sofia/internal/sip:60022 at 141.196.174.60:33822 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard HANGUP, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60022 at 141.196.174.60:33822) State HANGUP going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:2689 sofia/internal/60021 at my.server.ip.address skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60021 at my.server.ip.address) State EXECUTE going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_HANGUP 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60022 at 141.196.174.60:33822) State REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard REPORTING, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60022 at 141.196.174.60:33822) State REPORTING going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60021 at my.server.ip.address) State HANGUP 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:503 Channel sofia/internal/60021 at my.server.ip.address hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1499 Session 23 (sofia/internal/sip:60022 at 141.196.174.60:33822) Locked, Waiting on external entities 2013-05-11 12:14:14.980444 [NOTICE] switch_core_session.c:1517 Session 23 (sofia/internal/sip:60022 at 141.196.174.60:33822) Ended 2013-05-11 12:14:14.980444 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:60022 at 141.196.174.60:33822 [CS_DESTROY] 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:60022 at 141.196.174.60:33822) Callstate Change HANGUP -> DOWN 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:48 sofia/internal/60021 at my.server.ip.address Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60021 at my.server.ip.address) State HANGUP going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/60021 at my.server.ip.address) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60021 at my.server.ip.address) State REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60022 at 141.196.174.60:33822) State DESTROY 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:396 sofia/internal/ sip:60022 at 141.196.174.60:33822 SOFIA DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60022 at 141.196.174.60:33822) State DESTROY going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:92 sofia/internal/60021 at my.server.ip.address Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60021 at my.server.ip.address) State REPORTING going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/60021 at my.server.ip.address) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1499 Session 22 (sofia/internal/60021 at my.server.ip.address) Locked, Waiting on external entities 2013-05-11 12:14:15.000444 [NOTICE] switch_core_session.c:1517 Session 22 (sofia/internal/60021 at my.server.ip.address) Ended 2013-05-11 12:14:15.000444 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/60021 at my.server.ip.address [CS_DESTROY] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/60021 at my.server.ip.address) Callstate Change HANGUP -> DOWN 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60021 at my.server.ip.address) State DESTROY 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:396 sofia/internal/60021 at my.server.ip.address SOFIA DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:99 sofia/internal/60021 at my.server.ip.address Standard DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60021 at my.server.ip.address) State DESTROY going to sleep 2013-05-11 12:29:03.100449 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:1683 Register: From: [60022 at my.server.ip.address] Contact: ["user" ] Expires: [900] third test 2013-05-11 14:08:17.980434 [NOTICE] switch_channel.c:968 New Channel sofia/internal/60022 at my.server.ip.address[156488c0-ba2b-11e2-ad18-bda7cbfd9554] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_NEW 2013-05-11 14:08:17.980434 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/60022 at my.server.ip.address) State NEW 2013-05-11 14:08:18.000438 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 14:08:18.000438 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 14:08:18.000438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.000438 [DEBUG] sofia.c:1719 detaching session 156488c0-ba2b-11e2-ad18-bda7cbfd9554 2013-05-11 14:08:18.120438 [DEBUG] sofia.c:1811 Re-attaching to session 156488c0-ba2b-11e2-ad18-bda7cbfd9554 2013-05-11 14:08:18.120438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.120438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5578 Channel sofia/internal/60022 at my.server.ip.address entering state [received][100] 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 3577259178 3577259178 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4000 RTP/AVP 97 101 c=IN IP4 141.196.174.60 a=rtpmap:97 SILK/16000 a=fmtp:97 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 192.168.43.193 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+8RV0fxAu+3s1Fc8BQxiMa9ras5u/JDmQ9uCVagu a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CQ25iu0Z418+mKdV+nRcwXqkd5z+gUxuXsqQt40P 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5802 (sofia/internal/60022 at my.server.ip.address) State Change CS_NEW -> CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60022 at my.server.ip.address) State INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:86 sofia/internal/60022 at my.server.ip.address SOFIA INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:126 (sofia/internal/60022 at my.server.ip.address) State Change CS_INIT -> CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60022 at my.server.ip.address) State INIT going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:2003 (sofia/internal/60022 at my.server.ip.address) Callstate Change DOWN -> RINGING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60022 at my.server.ip.address) State ROUTING 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:149 sofia/internal/60022 at my.server.ip.address SOFIA ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:117 sofia/internal/60022 at my.server.ip.address Standard ROUTING 2013-05-11 14:08:18.140456 [INFO] mod_dialplan_xml.c:557 Processing 60022 <60022>->60021 in context default Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->unloop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->tod_example] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->holiday_example] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->global-intercept] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global-intercept] destination_number(60021) =~ /^886$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->group-intercept] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [group-intercept] destination_number(60021) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [intercept-ext] destination_number(60021) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->redial] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [redial] destination_number(60021) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->global] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 3577259178 3577259178 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4000 RTP/AVP 97 101 c=IN IP4 141.196.174.60 a=rtpmap:97 SILK/16000 a=fmtp:97 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 192.168.43.193 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+8RV0fxAu+3s1Fc8BQxiMa9ras5u/JDmQ9uCVagu a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CQ25iu0Z418+mKdV+nRcwXqkd5z+gUxuXsqQt40P ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Action set(sip_secure_media=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action export(sip_secure_media=true) Dialplan: sofia/internal/60022 at my.server.ip.address Absolute Condition [global] Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [snom-demo-2] destination_number(60021) =~ /^9001$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [snom-demo-1] destination_number(60021) =~ /^9000$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60021) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60021) =~ /^779$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call_return] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call_return] destination_number(60021) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->del-group] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [del-group] destination_number(60021) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->add-group] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [add-group] destination_number(60021) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call-group-simo] destination_number(60021) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call-group-order] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call-group-order] destination_number(60021) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [extension-intercom] destination_number(60021) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [Local_Extension] destination_number(60021) =~ /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Action export(dialed_extension=60021) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(ringback=${us-ring}) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(call_timeout=30) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(hangup_after_bridge=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(continue_on_fail=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/60022 at my.server.ip.address Action answer() Dialplan: sofia/internal/60022 at my.server.ip.address Action sleep(1000) Dialplan: sofia/internal/60022 at my.server.ip.address Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/60022 at my.server.ip.address) State Change CS_ROUTING -> CS_EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60022 at my.server.ip.address) State ROUTING going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60022 at my.server.ip.address) State EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:242 sofia/internal/60022 at my.server.ip.address SOFIA EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:209 sofia/internal/60022 at my.server.ip.address Standard EXECUTE EXECUTE sofia/internal/60022 at my.server.ip.addressset(sip_secure_media=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [sip_secure_media]=[true] EXECUTE sofia/internal/60022 at my.server.ip.addressexport(sip_secure_media=true) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_secure_media]=[true] EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-spymap/60022/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/60022/60021) EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/global/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address export(RFC2822_DATE=Sat, 11 May 2013 14:08:18 +0300) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] EXECUTE sofia/internal/60022 at my.server.ip.addressexport(dialed_extension=60021) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [dialed_extension]=[60021] EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(1 b s execute_extension::dx XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/60022.2013-05-11-14-08-18.wav) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/60022.2013-05-11-14-08-18.wav EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(3 b s execute_extension::cf XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(4 b s execute_extension::att_xfer XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/60022 at my.server.ip.addressset(ringback=%(2000,4000,440,480)) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/60022 at my.server.ip.addressset(transfer_ringback=local_stream://moh) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/60022 at my.server.ip.address set(call_timeout=30) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [call_timeout]=[30] EXECUTE sofia/internal/60022 at my.server.ip.addressset(hangup_after_bridge=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/60022 at my.server.ip.addressset(continue_on_fail=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [continue_on_fail]=[true] EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-call_return/60021/60022) EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/60021/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.addressset(called_party_callgroup=techsupport) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/techsupport/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial_ext/global/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.addresshash(insert/my.server.ip.address-last_dial/techsupport/156488c0-ba2b-11e2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.addressbridge(user/60021 at my.server.ip.address) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60021] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60021] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 14:08:18.140456 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[my.server.ip.address] 2013-05-11 14:08:18.140456 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[60021 at my.server.ip.address] 2013-05-11 14:08:18.140456 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:60021 at 141.196.174.60:57938[157eb0a6-ba2b-11e2-ad38-bda7cbfd9554] 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:4961 (sofia/internal/ sip:60021 at 141.196.174.60:57938) State Change CS_NEW -> CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:5031 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60021 at 141.196.174.60:57938) State INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:86 sofia/internal/ sip:60021 at 141.196.174.60:57938 SOFIA INIT 2013-05-11 14:08:18.140456 [DEBUG] sofia_glue.c:3157 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:FGwgO9qNK7dbHa/ZYQcA2fWT17ktjsvjEt5fYXf4] 2013-05-11 14:08:18.140456 [DEBUG] sofia_glue.c:2649 Local SDP: v=0 o=FreeSWITCH 1368243506 1368243507 IN IP4 my.server.ip.address s=FreeSWITCH c=IN IP4 my.server.ip.address t=0 0 m=audio 26992 RTP/SAVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FGwgO9qNK7dbHa/ZYQcA2fWT17ktjsvjEt5fYXf4 a=ptime:20 a=sendrecv m=audio 26992 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:126 (sofia/internal/ sip:60021 at 141.196.174.60:57938) State Change CS_INIT -> CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60021 at 141.196.174.60:57938) State INIT going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:2003 (sofia/internal/ sip:60021 at 141.196.174.60:57938) Callstate Change DOWN -> RINGING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60021 at 141.196.174.60:57938) State ROUTING 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:149 sofia/internal/ sip:60021 at 141.196.174.60:57938 SOFIA ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60021 at 141.196.174.60:57938) State ROUTING going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60021 at 141.196.174.60:57938) State CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60021 at 141.196.174.60:57938) State CONSUME_MEDIA going to sleep 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5578 Channel sofia/internal/ sip:60021 at 141.196.174.60:57938 entering state [calling][0] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] sofia.c:5578 Channel sofia/internal/ sip:60021 at 141.196.174.60:57938 entering state [terminated][406] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:2994 (sofia/internal/ sip:60021 at 141.196.174.60:57938) Callstate Change RINGING -> HANGUP 2013-05-11 14:08:18.340438 [NOTICE] sofia.c:6385 Hangup sofia/internal/ sip:60021 at 141.196.174.60:57938 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [KILL] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_HANGUP 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60021 at 141.196.174.60:57938) State HANGUP 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:503 Channel sofia/internal/ sip:60021 at 141.196.174.60:57938 hanging up, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60021 at 141.196.174.60:57938) State HANGUP going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60021 at 141.196.174.60:57938) State REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60021 at 141.196.174.60:57938) State REPORTING going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1499 Session 27 (sofia/internal/sip:60021 at 141.196.174.60:57938) Locked, Waiting on external entities 2013-05-11 14:08:18.340438 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1517 Session 27 (sofia/internal/sip:60021 at 141.196.174.60:57938) Ended 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:60021 at 141.196.174.60:57938 [CS_DESTROY] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:60021 at 141.196.174.60:57938) Callstate Change HANGUP -> DOWN 2013-05-11 14:08:18.340438 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60021 at 141.196.174.60:57938) State DESTROY 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:396 sofia/internal/ sip:60021 at 141.196.174.60:57938 SOFIA DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard DESTROY 2013-05-11 14:08:18.340438 [INFO] mod_dptools.c:3060 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60021 at 141.196.174.60:57938) State DESTROY going to sleep EXECUTE sofia/internal/60022 at my.server.ip.address answer() 2013-05-11 14:08:18.340438 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, refer to rfc3711 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:2994 (sofia/internal/60022 at my.server.ip.address) Callstate Change RINGING -> HANGUP 2013-05-11 14:08:18.340438 [NOTICE] switch_channel.c:3484 Hangup sofia/internal/60022 at my.server.ip.address [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/60022 at my.server.ip.address [KILL] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:2689 sofia/internal/60022 at my.server.ip.address skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60022 at my.server.ip.address) State EXECUTE going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_HANGUP 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60022 at my.server.ip.address) State HANGUP 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:497 sofia/internal/60022 at my.server.ip.address Overriding SIP cause 488 with 406 from the other leg 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:503 Channel sofia/internal/60022 at my.server.ip.address hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 406 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:48 sofia/internal/60022 at my.server.ip.address Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60022 at my.server.ip.address) State HANGUP going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/60022 at my.server.ip.address) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60022 at my.server.ip.address) State REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:92 sofia/internal/60022 at my.server.ip.address Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60022 at my.server.ip.address) State REPORTING going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/60022 at my.server.ip.address) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1499 Session 26 (sofia/internal/60022 at my.server.ip.address) Locked, Waiting on external entities 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1517 Session 26 (sofia/internal/60022 at my.server.ip.address) Ended 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/60022 at my.server.ip.address [CS_DESTROY] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/60022 at my.server.ip.address) Callstate Change HANGUP -> DOWN 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60022 at my.server.ip.address) State DESTROY 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:396 sofia/internal/60022 at my.server.ip.address SOFIA DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:99 sofia/internal/60022 at my.server.ip.address Standard DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60022 at my.server.ip.address) State DESTROY going to sleep 2013/5/10 Steven Ayre > Are you using the default dialplan? > > Chances are your destination_number condition's regex for the extension > that calls users is limited to 4 digits. > > Eg > \d\d\d\d > \d{4} > > -Steve > > > > On 10 May 2013 12:30, Burak BorYaz?l?m wrote: > >> Hello, >> >> I have problems with adding new user to freeswitch. When trying to add >> user with user id has different number of digits than 4, it can register >> but it cant call or cant be called. I changed local extension regular >> expression in dialpan default.xml but the changes only work with four >> digits users(user ids or dial number) .So I want dial a number that has >> five or more digits. What other configurations I must change. >> >> Thank you... >> Burak, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130511/f0767ea2/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sat May 11 15:58:13 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 11 May 2013 12:58:13 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> Message-ID: Hi Chris, There is a couple of things you can do here to determine what is happening. First, make sure you are running the latest freeswitch stable release. Enable console logging and set the log level to debug (console loglevel debug) iirc, Next, see if you are able to type 'status' into the CLI after it has stopped responding, and make sure it outputs something. Perform a test to see how long it takes for the freeswitch instance to stop responding. Copy all the log entries from the point where it started, to the point where it stopped responding, then attach and send the file to this list. Once the fs instance has stopped responding, can you please try and construct a SIP packet and send it manually to the server from the local host of the server. You can use sendip to do this, by following the instructions at [1]. Once you send the packet, check to see if it is received by fs. If no packet is being received even when sending from localhost, run "netstat -inp | grep LISTEN | SIPPORTHERE", replacing SIPPORTHERE with the port you are running sip on fs. Ensure that the only entry for udp is th fs instance, and nothing else. For a slightly more deeper approach, attach an strafe to the fs instance, you can do this by looking in the process list for the fs instance, then typing "strafe -ifq -p PIDHERE" replacing PIDHERE with the pid of your fs instance. Attempt to send a call to fs, ensure the packet arrives through wireshark, and see if the packet appears in the strace (this is a really brutal way of debugging sometimes, it doesn't offer all the answers but can give you a bit of an X-ray as to what is going on). Also ensure you eliminate any possible outside interference, such as switch gear or routing equipment.. If possible, connect the windows server and the fs instance directly together, bypassing anything in between. Ensure you also disable any local firewall on the fs server, I.e. iptables. This should hopefully give you a nudge in the right direction to determine where this problem is happening. Hope this helps! Cal ps) typed on my ipad, sorry for any spelling mistakes [1] http://www.moythreads.com/wordpress/2012/03/15/sending-udp-packets-from-the-command-line/ On Saturday, May 11, 2013, Christopher Hall wrote: > Steve,**** > > ** ** > > Thanks for your response, I have tried with global siptrace turned on and > I see absolutely nothing when I make the call, the SIP INVITE messages are > sent to the FreeSWITCH host machine as seen using WireShark running on that > host.**** > > ** ** > > Chris.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org 'cvml', 'freeswitch-users-bounces at lists.freeswitch.org');> [mailto: > freeswitch-users-bounces at lists.freeswitch.org 'freeswitch-users-bounces at lists.freeswitch.org');>] *On Behalf Of *Steven > Ayre > *Sent:* 10 May 2013 20:01 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE*** > * > > ** ** > > Try enabling the sip trace when it happens to see if FS receives the > packets:**** > > sofia global siptrace on**** > > ** ** > > Steve**** > > ** ** > > ** ** > > > On 10 May 2013, at 17:03, "Christopher Hall" > > wrote:**** > > Hi,**** > > **** > > I?m new to FreeSWITCH lists and I?m not sure if this is the correct list > for my post, any help will be much appreciated.**** > > **** > > I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) > SIP service. This works correctly and I can dial in and out using the > service. **** > > **** > > I have an application running on a Windows Server that interfaces using > event socket to the FreeSWITCH server. The intended use is that calls > dialling in through the draytel service are automatically handled by this > application. Again this works correctly.**** > > **** > > However, after a period of time which does not yet appear to be > deterministic the FreeSWITCH server stops responding to INVITE requests > from the draytel server. The FreeSWITCH server can still respond as I can > dial using a local network VOIP phone and the application handles that > call, but on the draytel connection there is no response.**** > > **** > > When FreeSWITCH stops responding I see absolutely nothing in the output > logging information. I have run WireShark on the server that is running > FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the > FreeSWITCH host but I don?t see any kind of response.**** > > **** > > Help!**** > > **** > > Thanks**** > > **** > > Chris.**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org 'consulting at freeswitch.org');> > http://www.freeswitchsolutions.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 'FreeSWITCH-users at 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/20130511/7e5f4d99/attachment.html From cal.leeming at simplicitymedialtd.co.uk Sat May 11 16:02:48 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sat, 11 May 2013 13:02:48 +0100 Subject: [Freeswitch-users] An strange bug on starting up FreeSwitch, any solution? In-Reply-To: References: Message-ID: Hello, Thanks for reporting this. Can you please follow these instructions, and post the results into a JIRA ticket. http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB This will allow us to see what is happening, and once it's in JIRA a core dev will be able to take a look. Cal On Saturday, May 11, 2013, Sayyed Mohammad Emami Razavi wrote: > after 10 seconds FS goes down! > below is my syslogs: > > [root at Ctel freeswitch]# tail -f /var/log/messages > May 11 18:55:00 aratel kernel: freeswitch[11261]: segfault at 1 ip > 00828282 sp b70539e8 error 4 in libc-2.12.so[6f7000+189000] > May 11 18:55:00 aratel abrt[11262]: saved core dump of pid 11193 > (/usr/local/freeswitch/bin/freeswitch) to > /var/spool/abrt/ccpp-2013-05-11-18:55:00-11193.new/coredump (8544256 bytes) > May 11 18:55:00 aratel abrtd: Directory 'ccpp-2013-05-11-18:55:00-11193' > creation detected > May 11 18:55:00 aratel abrtd: Executable > '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package > May 11 18:55:00 aratel abrtd: Corrupted or bad dump > /var/spool/abrt/ccpp-2013-05-11-18:55:00-11193 (res:2), deleting > May 11 19:17:23 aratel kernel: freeswitch[11342]: segfault at 1 ip > 00828282 sp b6e859e8 error 4 in libc-2.12.so[6f7000+189000] > May 11 19:17:23 aratel abrt[11343]: saved core dump of pid 11330 > (/usr/local/freeswitch/bin/freeswitch) to > /var/spool/abrt/ccpp-2013-05-11-19:17:23-11330.new/coredump (8540160 bytes) > May 11 19:17:23 aratel abrtd: Directory 'ccpp-2013-05-11-19:17:23-11330' > creation detected > May 11 19:17:23 aratel abrtd: Executable > '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package > May 11 19:17:23 aratel abrtd: Corrupted or bad dump > /var/spool/abrt/ccpp-2013-05-11-19:17:23-11330 (res:2), deleting > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130511/d700b63a/attachment.html From anthony.minessale at gmail.com Sat May 11 19:07:49 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 May 2013 10:07:49 -0500 Subject: [Freeswitch-users] An strange bug on starting up FreeSwitch, any solution? In-Reply-To: References: Message-ID: I'd do "make sure" first too for good measure. On Sat, May 11, 2013 at 7:02 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello, > > Thanks for reporting this. > > Can you please follow these instructions, and post the results into a JIRA > ticket. > > http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB > > This will allow us to see what is happening, and once it's in JIRA a core > dev will be able to take a look. > > Cal > > > On Saturday, May 11, 2013, Sayyed Mohammad Emami Razavi wrote: > >> after 10 seconds FS goes down! >> below is my syslogs: >> >> [root at Ctel freeswitch]# tail -f /var/log/messages >> May 11 18:55:00 aratel kernel: freeswitch[11261]: segfault at 1 ip >> 00828282 sp b70539e8 error 4 in libc-2.12.so[6f7000+189000] >> May 11 18:55:00 aratel abrt[11262]: saved core dump of pid 11193 >> (/usr/local/freeswitch/bin/freeswitch) to /var/spool/abrt/ccpp- >> 2013-05-11-18:55:00-11193.new/coredump (8544256 bytes) >> May 11 18:55:00 aratel abrtd: Directory 'ccpp-2013-05-11-18:55:00-11193' >> creation detected >> May 11 18:55:00 aratel abrtd: Executable >> '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package >> May 11 18:55:00 aratel abrtd: Corrupted or bad dump /var/spool/abrt/ccpp- >> 2013-05-11-18:55:00-11193 (res:2), deleting >> May 11 19:17:23 aratel kernel: freeswitch[11342]: segfault at 1 ip >> 00828282 sp b6e859e8 error 4 in libc-2.12.so[6f7000+189000] >> May 11 19:17:23 aratel abrt[11343]: saved core dump of pid 11330 >> (/usr/local/freeswitch/bin/freeswitch) to /var/spool/abrt/ccpp- >> 2013-05-11-19:17:23-11330.new/coredump (8540160 bytes) >> May 11 19:17:23 aratel abrtd: Directory 'ccpp-2013-05-11-19:17:23-11330' >> creation detected >> May 11 19:17:23 aratel abrtd: Executable >> '/usr/local/freeswitch/bin/freeswitch' doesn't belong to any package >> May 11 19:17:23 aratel abrtd: Corrupted or bad dump /var/spool/abrt/ccpp- >> 2013-05-11-19:17:23-11330 (res:2), deleting >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130511/e136ca39/attachment-0001.html From chris.hall at vividapps.co.uk Sat May 11 19:31:45 2013 From: chris.hall at vividapps.co.uk (Christopher Hall) Date: Sat, 11 May 2013 16:31:45 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> Message-ID: <017e01ce4e5c$a56a06b0$f03e1410$@vividapps.co.uk> Cal, Thanks for your response. I will perform as many of those tests as I can, but while I'm doing that I need to clarify my situation. On re-reading my original post I realise I have not been clear. . My FS is running on Windows as a service. . FS is running on the same virtual host as the automated call handling application. . FS does not stop responding completely but only to SIP INVITE from the draytel service . When FS is in this state I can still call the automated call handling application using a soft phone on my LAN. . When FS is in this state I do not see any log output from FS when the draytel service sends a SIP INVITE. I hope that is more clear, please let me know if any of the above alters your thinking. I will attempt to work through your suggestions. Chris. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Cal Leeming [Simplicity Media Ltd] Sent: 11 May 2013 12:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE Hi Chris, There is a couple of things you can do here to determine what is happening. First, make sure you are running the latest freeswitch stable release. Enable console logging and set the log level to debug (console loglevel debug) iirc, Next, see if you are able to type 'status' into the CLI after it has stopped responding, and make sure it outputs something. Perform a test to see how long it takes for the freeswitch instance to stop responding. Copy all the log entries from the point where it started, to the point where it stopped responding, then attach and send the file to this list. Once the fs instance has stopped responding, can you please try and construct a SIP packet and send it manually to the server from the local host of the server. You can use sendip to do this, by following the instructions at [1]. Once you send the packet, check to see if it is received by fs. If no packet is being received even when sending from localhost, run "netstat -inp | grep LISTEN | SIPPORTHERE", replacing SIPPORTHERE with the port you are running sip on fs. Ensure that the only entry for udp is th fs instance, and nothing else. For a slightly more deeper approach, attach an strafe to the fs instance, you can do this by looking in the process list for the fs instance, then typing "strafe -ifq -p PIDHERE" replacing PIDHERE with the pid of your fs instance. Attempt to send a call to fs, ensure the packet arrives through wireshark, and see if the packet appears in the strace (this is a really brutal way of debugging sometimes, it doesn't offer all the answers but can give you a bit of an X-ray as to what is going on). Also ensure you eliminate any possible outside interference, such as switch gear or routing equipment.. If possible, connect the windows server and the fs instance directly together, bypassing anything in between. Ensure you also disable any local firewall on the fs server, I.e. iptables. This should hopefully give you a nudge in the right direction to determine where this problem is happening. Hope this helps! Cal ps) typed on my ipad, sorry for any spelling mistakes [1] http://www.moythreads.com/wordpress/2012/03/15/sending-udp-packets-from-the- command-line/ On Saturday, May 11, 2013, Christopher Hall wrote: Steve, Thanks for your response, I have tried with global siptrace turned on and I see absolutely nothing when I make the call, the SIP INVITE messages are sent to the FreeSWITCH host machine as seen using WireShark running on that host. Chris. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Steven Ayre Sent: 10 May 2013 20:01 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE Try enabling the sip trace when it happens to see if FS receives the packets: sofia global siptrace on Steve On 10 May 2013, at 17:03, "Christopher Hall" > wrote: Hi, I'm new to FreeSWITCH lists and I'm not sure if this is the correct list for my post, any help will be much appreciated. I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) SIP service. This works correctly and I can dial in and out using the service. I have an application running on a Windows Server that interfaces using event socket to the FreeSWITCH server. The intended use is that calls dialling in through the draytel service are automatically handled by this application. Again this works correctly. However, after a period of time which does not yet appear to be deterministic the FreeSWITCH server stops responding to INVITE requests from the draytel server. The FreeSWITCH server can still respond as I can dial using a local network VOIP phone and the application handles that call, but on the draytel connection there is no response. When FreeSWITCH stops responding I see absolutely nothing in the output logging information. I have run WireShark on the server that is running FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the FreeSWITCH host but I don't see any kind of response. Help! Thanks Chris. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130511/d27d690e/attachment-0001.html From avi at avimarcus.net Sat May 11 21:39:34 2013 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 11 May 2013 20:39:34 +0300 Subject: [Freeswitch-users] Caller ID - a moving target In-Reply-To: <138701ce4de7$4fad61b0$ef082510$@bizfocused.com> References: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> <138701ce4de7$4fad61b0$ef082510$@bizfocused.com> Message-ID: SIP includes both a name and a number. It seems each ATA/phone has their own manner of how they choose to display it. (Hint: if there's only 1 line on the LCD, you'll probably be disappointed in what it chooses...) -Avi On Sat, May 11, 2013 at 4:31 AM, Sean Devoy wrote: > Thank you Jo?o, but you don?t get off that easy!**** > > ** ** > > Can I get CID Name *and* Number displayed in either/both cases?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o > Mesquita > *Sent:* Friday, May 10, 2013 4:29 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Caller ID - a moving target**** > > ** ** > > ignore_display_updates=true will disable this behavior. That's a FS thing > and it is supposed to do that as SIP supports cid updates.**** > > > **** > > Jo?o Mesquita > FreeSWITCH? Solutions**** > > ** ** > > On Fri, May 10, 2013 at 5:01 PM, Sean Devoy wrote: > **** > > **** > > Hi,**** > > **** > > When someone dials my extension from an internal extension, my phone > displays the CID Number. When I answer, the display switches to CID Name.. > **** > > **** > > Is that a ?feature? of Cisco SPA504G phones or FS?**** > > **** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130511/25e651b4/attachment.html From sdevoy at bizfocused.com Sat May 11 19:33:15 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 11 May 2013 11:33:15 -0400 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: Message-ID: <15f401ce4e5c$dae01780$90a04680$@bizfocused.com> I cannot tell you why this is only a problem with extensions w/ 5 digits, but I can tell you what failed here. Your underlying error is: 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, refer to rfc3711 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> HANGUP 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Can you try it without the crypto stuff and paste the same output to pastebin.freeswitch.org (not here in email)? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Burak BorYazilim Sent: Saturday, May 11, 2013 7:28 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch User Adding First off all thanks for your kind helps. Let me tell you the situation more clearly. Firstly I used Android phones as sip clients with the open source sip client program, CSipSimple. My server computer has Ubuntu 12.04 LTS with one static ip address. freeswitch version output: FreeSWITCH Version 1.3.13b+git~20130205T003128Z~70a9560306 (git 70a9560 2013-02-05 00:31:28Z) We were using the system with 4 digits number sip account names without any problem. Also the account name will be used as dial number. An example of the user xml file is below. I changed the regex in dialpan/default.xml. The changings were perfectly succesfull with, again, 4 digits number. But when trying to include any other number of digits (3, 5 and 6 were tested), it doesn't work(When I change the regex to accept only 5 digits numbers, 4 digits ones didnt work as expected). I really could not understand why it is happening. Why there was no problem with 4 digits number and why the exact same system does not work with this basic change. To explain the errors more, I want to talk about my tests. Firstly I used 4 digits number user. This test repeated with tls and srtp. These two test were succesfull. And the secand test is same with first test but with 5 digits numbers without tls and srtp. Registration was succesfull but cant call. (or the call could not be forwarded) Of course I changed the regex and execute reloadxml in this test. GSM (8kHz) and G722(16kHz) codecs were used. As you see below, CODEC NEGOTIATION ERROR occured. After getting this error I changed the codes as SILK(16kHz). With this change I made the third test. Third test result is also below. show registretions output: reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostn ame,metadata 60022,my.server.ip.address,oOnxiVJFAgQWLABrD01UsYTVOY3TVSlx,sofia/internal/s ip:60022 at 141.196.174.60:52245;ob,1368271435,141.196.174.60,52245,udp,server, 60021,my.server.ip.address,5r4MTDiZPhs5qzdin9A3hEUh1zZsdqqk,sofia/internal/s ip:60021 at 141.196.174.60:57938;ob,1368271446,141.196.174.60,57938,udp,server, @Sean, sorry but I cant post my whole log file because my server has got a network attack so I am quite busy with this attacker. And also I dont think CSipSimple has a problem with 5 digits because this system (with 5 digits) was succesfull with Kamailio and CSipSimple. If I could not be clear please let me know. Thanks... user xml example: This was exact same with the default users in freeswitch. Only the number changed. The second test error: 2013-05-11 12:13:56.800443 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 12:13:56.900444 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:1683 Register: From: [60021 at my.server.ip.address] Contact: ["user" ] Expires: [900] 2013-05-11 12:14:06.820443 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:06.900444 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:1683 Register: From: [60022 at my.server.ip.address] Contact: ["user" ] Expires: [900] 2013-05-11 12:14:14.640437 [NOTICE] switch_channel.c:968 New Channel sofia/internal/60021 at my.server.ip.address [26730cd2-ba1b-11e2-acc5-bda7cbfd9554] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_NEW 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/60021 at my.server.ip.address) State NEW 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 12:14:14.660438 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:14:14.660438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:1719 detaching session 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 2013-05-11 12:14:14.780439 [DEBUG] sofia.c:1811 Re-attaching to session 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60021' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel sofia/internal/60021 at my.server.ip.address entering state [received][100] 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5802 (sofia/internal/60021 at my.server.ip.address) State Change CS_NEW -> CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60021 at my.server.ip.address) State INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 sofia/internal/60021 at my.server.ip.address SOFIA INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 (sofia/internal/60021 at my.server.ip.address) State Change CS_INIT -> CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60021 at my.server.ip.address) State INIT going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 (sofia/internal/60021 at my.server.ip.address) Callstate Change DOWN -> RINGING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60021 at my.server.ip.address) State ROUTING 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 sofia/internal/60021 at my.server.ip.address SOFIA ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:117 sofia/internal/60021 at my.server.ip.address Standard ROUTING 2013-05-11 12:14:14.800439 [INFO] mod_dialplan_xml.c:557 Processing 60021 <60021>->60022 in context default Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->unloop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->tod_example] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->holiday_example] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->global-intercept] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global-intercept] destination_number(60022) =~ /^886$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->group-intercept] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [group-intercept] destination_number(60022) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [intercept-ext] destination_number(60022) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->redial] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [redial] destination_number(60022) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->global] continue=true Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/60021 at my.server.ip.address Action set(sip_secure_media=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action export(sip_secure_media=true) Dialplan: sofia/internal/60021 at my.server.ip.address Absolute Condition [global] Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [snom-demo-2] destination_number(60022) =~ /^9001$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [snom-demo-1] destination_number(60022) =~ /^9000$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60022) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60022) =~ /^779$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call_return] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call_return] destination_number(60022) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->del-group] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [del-group] destination_number(60022) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->add-group] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [add-group] destination_number(60022) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call-group-simo] destination_number(60022) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->call-group-order] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [call-group-order] destination_number(60022) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [extension-intercom] destination_number(60022) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [Local_Extension] destination_number(60022) =~ /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/60021 at my.server.ip.address Action export(dialed_extension=60022) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(ringback=${us-ring}) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(call_timeout=30) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(hangup_after_bridge=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(continue_on_fail=true) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60021 at my.server.ip.address Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/60021 at my.server.ip.address Action answer() Dialplan: sofia/internal/60021 at my.server.ip.address Action sleep(1000) Dialplan: sofia/internal/60021 at my.server.ip.address Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/60021 at my.server.ip.address) State Change CS_ROUTING -> CS_EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60021 at my.server.ip.address) State ROUTING going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60021 at my.server.ip.address) State EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:242 sofia/internal/60021 at my.server.ip.address SOFIA EXECUTE 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:209 sofia/internal/60021 at my.server.ip.address Standard EXECUTE EXECUTE sofia/internal/60021 at my.server.ip.address set(sip_secure_media=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [sip_secure_media]=[true] EXECUTE sofia/internal/60021 at my.server.ip.address export(sip_secure_media=true) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_secure_media]=[true] EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-spymap/60021/26730cd2-ba1b-11e2-acc5-bda7cb fd9554) EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/60021/60022) EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/global/26730cd2-ba1b-11e2-acc5-bd a7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address export(RFC2822_DATE=Sat, 11 May 2013 12:14:14 +0300) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] EXECUTE sofia/internal/60021 at my.server.ip.address export(dialed_extension=60022) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [dialed_extension]=[60022] EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(1 b s execute_extension::dx XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/60021.2013-05-11-12-14-14.w av) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/60021.2013-05-11-12-14-14.w av EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(3 b s execute_extension::cf XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/60021 at my.server.ip.address bind_meta_app(4 b s execute_extension::att_xfer XML features) 2013-05-11 12:14:14.800439 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/60021 at my.server.ip.address set(ringback=%(2000,4000,440,480)) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/60021 at my.server.ip.address set(transfer_ringback=local_stream://moh) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/60021 at my.server.ip.address set(call_timeout=30) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [call_timeout]=[30] EXECUTE sofia/internal/60021 at my.server.ip.address set(hangup_after_bridge=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/60021 at my.server.ip.address set(continue_on_fail=true) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [continue_on_fail]=[true] EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-call_return/60022/60021) EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/60022/26730cd2-ba1b-11e2-acc5 -bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address set(called_party_callgroup=techsupport) 2013-05-11 12:14:14.800439 [DEBUG] mod_dptools.c:1349 sofia/internal/60021 at my.server.ip.address SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/techsupport/26730cd2-ba1b-11e 2-acc5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/global/26730cd2-ba1b-11e2-acc 5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/techsupport/26730cd2-ba1b-11e2-ac c5-bda7cbfd9554) EXECUTE sofia/internal/60021 at my.server.ip.address bridge(user/60022 at my.server.ip.address) 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60022] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 12:14:14 +0300] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:1089 sofia/internal/60021 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60022] to event 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 12:14:14.800439 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[my.server.ip.address] 2013-05-11 12:14:14.800439 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[60022 at my.server.ip.address] 2013-05-11 12:14:14.800439 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:60022 at 141.196.174.60:33822 [268c95ee-ba1b-11e2-ace5-bda7cbfd9554] 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:4961 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_NEW -> CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:5031 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:5034 [zrtp_passthru] Setting b-leg absolute_codec_string='PCMA at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,GSM at 8000h @20i at 13200b' 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_INIT 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60022 at 141.196.174.60:33822) State INIT 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 sofia/internal/sip:60022 at 141.196.174.60:33822 SOFIA INIT 2013-05-11 12:14:14.800439 [DEBUG] sofia_glue.c:3157 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:jy7Mnu44PUrnS4nFSUGkaIsFNftmZRTnE61m4sui] 2013-05-11 12:14:14.800439 [DEBUG] sofia_glue.c:2649 Local SDP: v=0 o=FreeSWITCH 1368242802 1368242803 IN IP4 my.server.ip.address s=FreeSWITCH c=IN IP4 my.server.ip.address t=0 0 m=audio 20852 RTP/SAVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:jy7Mnu44PUrnS4nFSUGkaIsFNftmZRTnE61m4sui a=ptime:20 a=sendrecv m=audio 20852 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_INIT -> CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60022 at 141.196.174.60:33822) State INIT going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:60022 at 141.196.174.60:33822) Callstate Change DOWN -> RINGING 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60022 at 141.196.174.60:33822) State ROUTING 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 sofia/internal/sip:60022 at 141.196.174.60:33822 SOFIA ROUTING 2013-05-11 12:14:14.800439 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60022 at 141.196.174.60:33822) State ROUTING going to sleep 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60022 at 141.196.174.60:33822) State CONSUME_MEDIA 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60022 at 141.196.174.60:33822) State CONSUME_MEDIA going to sleep 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel sofia/internal/sip:60022 at 141.196.174.60:33822 entering state [calling][0] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] sofia.c:5578 Channel sofia/internal/sip:60022 at 141.196.174.60:33822 entering state [proceeding][180] 2013-05-11 12:14:14.980444 [NOTICE] sofia.c:5670 Ring-Ready sofia/internal/sip:60022 at 141.196.174.60:33822! 2013-05-11 12:14:14.980444 [INFO] switch_ivr_originate.c:1185 Sending early media 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, refer to rfc3711 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 3577252345 3577252345 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4010 RTP/AVP 8 0 3 101 c=IN IP4 141.196.174.60 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4011 IN IP4 192.168.43.10 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> HANGUP 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/60021 at my.server.ip.address [KILL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:1186 sofia/internal/60021 at my.server.ip.address Media Establishment Failed. 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:60022 at 141.196.174.60:33822) Callstate Change RINGING -> HANGUP 2013-05-11 12:14:14.980444 [NOTICE] switch_ivr_originate.c:3620 Hangup sofia/internal/sip:60022 at 141.196.174.60:33822 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_HANGUP 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [KILL] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60022 at 141.196.174.60:33822) State HANGUP 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:60022 at 141.196.174.60:33822 hanging up, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2013-05-11 12:14:14.980444 [INFO] mod_dptools.c:3060 Originate Failed. Cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:562 Sending CANCEL to sofia/internal/sip:60022 at 141.196.174.60:33822 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard HANGUP, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60022 at 141.196.174.60:33822) State HANGUP going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:2689 sofia/internal/60021 at my.server.ip.address skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60021 at my.server.ip.address) State EXECUTE going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_HANGUP 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60022 at 141.196.174.60:33822) State REPORTING 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard REPORTING, cause: ORIGINATOR_CANCEL 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60022 at 141.196.174.60:33822) State REPORTING going to sleep 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60021 at my.server.ip.address) State HANGUP 2013-05-11 12:14:14.980444 [DEBUG] mod_sofia.c:503 Channel sofia/internal/60021 at my.server.ip.address hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:14.980444 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:60022 at 141.196.174.60:33822) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60022 at 141.196.174.60:33822 [BREAK] 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:1499 Session 23 (sofia/internal/sip:60022 at 141.196.174.60:33822) Locked, Waiting on external entities 2013-05-11 12:14:14.980444 [NOTICE] switch_core_session.c:1517 Session 23 (sofia/internal/sip:60022 at 141.196.174.60:33822) Ended 2013-05-11 12:14:14.980444 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:60022 at 141.196.174.60:33822 [CS_DESTROY] 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 488 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:60022 at 141.196.174.60:33822) Callstate Change HANGUP -> DOWN 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:48 sofia/internal/60021 at my.server.ip.address Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60021 at my.server.ip.address) State HANGUP going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/60021 at my.server.ip.address) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:60022 at 141.196.174.60:33822) Running State Change CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60021 at my.server.ip.address) State REPORTING 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60022 at 141.196.174.60:33822) State DESTROY 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:396 sofia/internal/sip:60022 at 141.196.174.60:33822 SOFIA DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:60022 at 141.196.174.60:33822 Standard DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60022 at 141.196.174.60:33822) State DESTROY going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:92 sofia/internal/60021 at my.server.ip.address Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60021 at my.server.ip.address) State REPORTING going to sleep 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/60021 at my.server.ip.address) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60021 at my.server.ip.address [BREAK] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_session.c:1499 Session 22 (sofia/internal/60021 at my.server.ip.address) Locked, Waiting on external entities 2013-05-11 12:14:15.000444 [NOTICE] switch_core_session.c:1517 Session 22 (sofia/internal/60021 at my.server.ip.address) Ended 2013-05-11 12:14:15.000444 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/60021 at my.server.ip.address [CS_DESTROY] 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/60021 at my.server.ip.address) Callstate Change HANGUP -> DOWN 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/60021 at my.server.ip.address) Running State Change CS_DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60021 at my.server.ip.address) State DESTROY 2013-05-11 12:14:15.000444 [DEBUG] mod_sofia.c:396 sofia/internal/60021 at my.server.ip.address SOFIA DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:99 sofia/internal/60021 at my.server.ip.address Standard DESTROY 2013-05-11 12:14:15.000444 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60021 at my.server.ip.address) State DESTROY going to sleep 2013-05-11 12:29:03.100449 [DEBUG] sofia_reg.c:1511 Send challenge for [60022 at my.server.ip.address] 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 12:29:05.900434 [DEBUG] sofia_reg.c:1683 Register: From: [60022 at my.server.ip.address] Contact: ["user" ] Expires: [900] third test 2013-05-11 14:08:17.980434 [NOTICE] switch_channel.c:968 New Channel sofia/internal/60022 at my.server.ip.address [156488c0-ba2b-11e2-ad18-bda7cbfd9554] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:17.980434 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_NEW 2013-05-11 14:08:17.980434 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/60022 at my.server.ip.address) State NEW 2013-05-11 14:08:18.000438 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 14:08:18.000438 [DEBUG] sofia_reg.c:1511 Send challenge for [60021 at my.server.ip.address] 2013-05-11 14:08:18.000438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.000438 [DEBUG] sofia.c:1719 detaching session 156488c0-ba2b-11e2-ad18-bda7cbfd9554 2013-05-11 14:08:18.120438 [DEBUG] sofia.c:1811 Re-attaching to session 156488c0-ba2b-11e2-ad18-bda7cbfd9554 2013-05-11 14:08:18.120438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.120438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected by acl "domains". Falling back to Digest auth. 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'record_stereo' = 'true' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_gateway' = 'example.com' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'default_areacode' = '918' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'transfer_fallback_extension' = 'operator' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'toll_allow' = 'domestic,international,local' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'accountcode' = '60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'user_context' = 'default' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_name' = 'Extension 60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'effective_caller_id_number' = '60022' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_name' = 'FreeSWITCH' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'outbound_caller_id_number' = '0000000000' 2013-05-11 14:08:18.140456 [DEBUG] sofia_reg.c:2767 event_add_header -> 'callgroup' = 'techsupport' 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5578 Channel sofia/internal/60022 at my.server.ip.address entering state [received][100] 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5589 Remote SDP: v=0 o=- 3577259178 3577259178 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4000 RTP/AVP 97 101 c=IN IP4 141.196.174.60 a=rtpmap:97 SILK/16000 a=fmtp:97 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 192.168.43.193 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+8RV0fxAu+3s1Fc8BQxiMa9ras5u/JDmQ9uCVagu a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CQ25iu0Z418+mKdV+nRcwXqkd5z+gUxuXsqQt40P 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5802 (sofia/internal/60022 at my.server.ip.address) State Change CS_NEW -> CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60022 at my.server.ip.address) State INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:86 sofia/internal/60022 at my.server.ip.address SOFIA INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:126 (sofia/internal/60022 at my.server.ip.address) State Change CS_INIT -> CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/60022 at my.server.ip.address) State INIT going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:2003 (sofia/internal/60022 at my.server.ip.address) Callstate Change DOWN -> RINGING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60022 at my.server.ip.address) State ROUTING 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:149 sofia/internal/60022 at my.server.ip.address SOFIA ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:117 sofia/internal/60022 at my.server.ip.address Standard ROUTING 2013-05-11 14:08:18.140456 [INFO] mod_dialplan_xml.c:557 Processing 60022 <60022>->60021 in context default Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->unloop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->tod_example] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->holiday_example] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->global-intercept] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global-intercept] destination_number(60021) =~ /^886$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->group-intercept] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [group-intercept] destination_number(60021) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [intercept-ext] destination_number(60021) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->redial] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [redial] destination_number(60021) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->global] continue=true Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [global] ${switch_r_sdp}(v=0 o=- 3577259178 3577259178 IN IP4 141.196.174.60 s=pjmedia c=IN IP4 141.196.174.60 t=0 0 m=audio 4000 RTP/AVP 97 101 c=IN IP4 141.196.174.60 a=rtpmap:97 SILK/16000 a=fmtp:97 useinbandfec=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4001 IN IP4 192.168.43.193 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+8RV0fxAu+3s1Fc8BQxiMa9ras5u/JDmQ9uCVagu a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:CQ25iu0Z418+mKdV+nRcwXqkd5z+gUxuXsqQt40P ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never Dialplan: sofia/internal/60022 at my.server.ip.address Action set(sip_secure_media=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action export(sip_secure_media=true) Dialplan: sofia/internal/60022 at my.server.ip.address Absolute Condition [global] Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [snom-demo-2] destination_number(60021) =~ /^9001$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [snom-demo-1] destination_number(60021) =~ /^9000$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60021) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [eavesdrop] destination_number(60021) =~ /^779$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call_return] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call_return] destination_number(60021) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->del-group] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [del-group] destination_number(60021) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->add-group] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [add-group] destination_number(60021) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call-group-simo] destination_number(60021) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->call-group-order] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [call-group-order] destination_number(60021) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (FAIL) [extension-intercom] destination_number(60021) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/60022 at my.server.ip.address Regex (PASS) [Local_Extension] destination_number(60021) =~ /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/60022 at my.server.ip.address Action export(dialed_extension=60021) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(ringback=${us-ring}) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(call_timeout=30) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(hangup_after_bridge=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(continue_on_fail=true) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/60022 at my.server.ip.address Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/60022 at my.server.ip.address Action answer() Dialplan: sofia/internal/60022 at my.server.ip.address Action sleep(1000) Dialplan: sofia/internal/60022 at my.server.ip.address Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/60022 at my.server.ip.address) State Change CS_ROUTING -> CS_EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/60022 at my.server.ip.address) State ROUTING going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60022 at my.server.ip.address) State EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:242 sofia/internal/60022 at my.server.ip.address SOFIA EXECUTE 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:209 sofia/internal/60022 at my.server.ip.address Standard EXECUTE EXECUTE sofia/internal/60022 at my.server.ip.address set(sip_secure_media=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [sip_secure_media]=[true] EXECUTE sofia/internal/60022 at my.server.ip.address export(sip_secure_media=true) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [sip_secure_media]=[true] EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-spymap/60022/156488c0-ba2b-11e2-ad18-bda7cb fd9554) EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/60022/60021) EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/global/156488c0-ba2b-11e2-ad18-bd a7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address export(RFC2822_DATE=Sat, 11 May 2013 14:08:18 +0300) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] EXECUTE sofia/internal/60022 at my.server.ip.address export(dialed_extension=60021) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1135 EXPORT (export_vars) [dialed_extension]=[60021] EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(1 b s execute_extension::dx XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/60022.2013-05-11-14-08-18.w av) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/60022.2013-05-11-14-08-18.w av EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(3 b s execute_extension::cf XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/60022 at my.server.ip.address bind_meta_app(4 b s execute_extension::att_xfer XML features) 2013-05-11 14:08:18.140456 [INFO] switch_ivr_async.c:3409 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/60022 at my.server.ip.address set(ringback=%(2000,4000,440,480)) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [ringback]=[%(2000,4000,440,480)] EXECUTE sofia/internal/60022 at my.server.ip.address set(transfer_ringback=local_stream://moh) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/60022 at my.server.ip.address set(call_timeout=30) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [call_timeout]=[30] EXECUTE sofia/internal/60022 at my.server.ip.address set(hangup_after_bridge=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/60022 at my.server.ip.address set(continue_on_fail=true) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [continue_on_fail]=[true] EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-call_return/60021/60022) EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/60021/156488c0-ba2b-11e2-ad18 -bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address set(called_party_callgroup=techsupport) 2013-05-11 14:08:18.140456 [DEBUG] mod_dptools.c:1349 sofia/internal/60022 at my.server.ip.address SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/techsupport/156488c0-ba2b-11e 2-ad18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial_ext/global/156488c0-ba2b-11e2-ad1 8-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address hash(insert/my.server.ip.address-last_dial/techsupport/156488c0-ba2b-11e2-ad 18-bda7cbfd9554) EXECUTE sofia/internal/60022 at my.server.ip.address bridge(user/60021 at my.server.ip.address) 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60021] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [sip_secure_media]=[true] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [RFC2822_DATE]=[Sat, 11 May 2013 14:08:18 +0300] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:1089 sofia/internal/60022 at my.server.ip.address EXPORTING[export_vars] [dialed_extension]=[60021] to event 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:2022 Parsing global variables 2013-05-11 14:08:18.140456 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[my.server.ip.address] 2013-05-11 14:08:18.140456 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[60021 at my.server.ip.address] 2013-05-11 14:08:18.140456 [NOTICE] switch_channel.c:968 New Channel sofia/internal/sip:60021 at 141.196.174.60:57938 [157eb0a6-ba2b-11e2-ad38-bda7cbfd9554] 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:4961 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_NEW -> CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:5031 [zrtp_passthru] Setting a-leg inherit_codec=true 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_INIT 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60021 at 141.196.174.60:57938) State INIT 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:86 sofia/internal/sip:60021 at 141.196.174.60:57938 SOFIA INIT 2013-05-11 14:08:18.140456 [DEBUG] sofia_glue.c:3157 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline:FGwgO9qNK7dbHa/ZYQcA2fWT17ktjsvjEt5fYXf4] 2013-05-11 14:08:18.140456 [DEBUG] sofia_glue.c:2649 Local SDP: v=0 o=FreeSWITCH 1368243506 1368243507 IN IP4 my.server.ip.address s=FreeSWITCH c=IN IP4 my.server.ip.address t=0 0 m=audio 26992 RTP/SAVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FGwgO9qNK7dbHa/ZYQcA2fWT17ktjsvjEt5fYXf4 a=ptime:20 a=sendrecv m=audio 26992 RTP/AVP 9 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:126 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_INIT -> CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:60021 at 141.196.174.60:57938) State INIT going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_channel.c:2003 (sofia/internal/sip:60021 at 141.196.174.60:57938) Callstate Change DOWN -> RINGING 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60021 at 141.196.174.60:57938) State ROUTING 2013-05-11 14:08:18.140456 [DEBUG] mod_sofia.c:149 sofia/internal/sip:60021 at 141.196.174.60:57938 SOFIA ROUTING 2013-05-11 14:08:18.140456 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:470 (sofia/internal/sip:60021 at 141.196.174.60:57938) State ROUTING going to sleep 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60021 at 141.196.174.60:57938) State CONSUME_MEDIA 2013-05-11 14:08:18.140456 [DEBUG] switch_core_state_machine.c:489 (sofia/internal/sip:60021 at 141.196.174.60:57938) State CONSUME_MEDIA going to sleep 2013-05-11 14:08:18.140456 [DEBUG] sofia.c:5578 Channel sofia/internal/sip:60021 at 141.196.174.60:57938 entering state [calling][0] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:975 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] sofia.c:5578 Channel sofia/internal/sip:60021 at 141.196.174.60:57938 entering state [terminated][406] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:2994 (sofia/internal/sip:60021 at 141.196.174.60:57938) Callstate Change RINGING -> HANGUP 2013-05-11 14:08:18.340438 [NOTICE] sofia.c:6385 Hangup sofia/internal/sip:60021 at 141.196.174.60:57938 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [KILL] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_HANGUP 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60021 at 141.196.174.60:57938) State HANGUP 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:503 Channel sofia/internal/sip:60021 at 141.196.174.60:57938 hanging up, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:48 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/sip:60021 at 141.196.174.60:57938) State HANGUP going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60021 at 141.196.174.60:57938) State REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:92 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/sip:60021 at 141.196.174.60:57938) State REPORTING going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/sip:60021 at 141.196.174.60:57938) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/sip:60021 at 141.196.174.60:57938 [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1499 Session 27 (sofia/internal/sip:60021 at 141.196.174.60:57938) Locked, Waiting on external entities 2013-05-11 14:08:18.340438 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1517 Session 27 (sofia/internal/sip:60021 at 141.196.174.60:57938) Ended 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/sip:60021 at 141.196.174.60:57938 [CS_DESTROY] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/sip:60021 at 141.196.174.60:57938) Callstate Change HANGUP -> DOWN 2013-05-11 14:08:18.340438 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_ivr_originate.c:3533 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/sip:60021 at 141.196.174.60:57938) Running State Change CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60021 at 141.196.174.60:57938) State DESTROY 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:396 sofia/internal/sip:60021 at 141.196.174.60:57938 SOFIA DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:99 sofia/internal/sip:60021 at 141.196.174.60:57938 Standard DESTROY 2013-05-11 14:08:18.340438 [INFO] mod_dptools.c:3060 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/sip:60021 at 141.196.174.60:57938) State DESTROY going to sleep EXECUTE sofia/internal/60022 at my.server.ip.address answer() 2013-05-11 14:08:18.340438 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, refer to rfc3711 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:830 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:2994 (sofia/internal/60022 at my.server.ip.address) Callstate Change RINGING -> HANGUP 2013-05-11 14:08:18.340438 [NOTICE] switch_channel.c:3484 Hangup sofia/internal/60022 at my.server.ip.address [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2013-05-11 14:08:18.340438 [DEBUG] switch_channel.c:3017 Send signal sofia/internal/60022 at my.server.ip.address [KILL] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:2689 sofia/internal/60022 at my.server.ip.address skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/60022 at my.server.ip.address) State EXECUTE going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_HANGUP 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60022 at my.server.ip.address) State HANGUP 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:497 sofia/internal/60022 at my.server.ip.address Overriding SIP cause 488 with 406 from the other leg 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:503 Channel sofia/internal/60022 at my.server.ip.address hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:633 Responding to INVITE with: 406 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:48 sofia/internal/60022 at my.server.ip.address Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:667 (sofia/internal/60022 at my.server.ip.address) State HANGUP going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:446 (sofia/internal/60022 at my.server.ip.address) State Change CS_HANGUP -> CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60022 at my.server.ip.address) State REPORTING 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:92 sofia/internal/60022 at my.server.ip.address Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/60022 at my.server.ip.address) State REPORTING going to sleep 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/60022 at my.server.ip.address) State Change CS_REPORTING -> CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1291 Send signal sofia/internal/60022 at my.server.ip.address [BREAK] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_session.c:1499 Session 26 (sofia/internal/60022 at my.server.ip.address) Locked, Waiting on external entities 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1517 Session 26 (sofia/internal/60022 at my.server.ip.address) Ended 2013-05-11 14:08:18.340438 [NOTICE] switch_core_session.c:1521 Close Channel sofia/internal/60022 at my.server.ip.address [CS_DESTROY] 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:556 (sofia/internal/60022 at my.server.ip.address) Callstate Change HANGUP -> DOWN 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/60022 at my.server.ip.address) Running State Change CS_DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60022 at my.server.ip.address) State DESTROY 2013-05-11 14:08:18.340438 [DEBUG] mod_sofia.c:396 sofia/internal/60022 at my.server.ip.address SOFIA DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:99 sofia/internal/60022 at my.server.ip.address Standard DESTROY 2013-05-11 14:08:18.340438 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/60022 at my.server.ip.address) State DESTROY going to sleep 2013/5/10 Steven Ayre Are you using the default dialplan? Chances are your destination_number condition's regex for the extension that calls users is limited to 4 digits. Eg \d\d\d\d \d{4} -Steve On 10 May 2013 12:30, Burak BorYaz?l?m wrote: Hello, I have problems with adding new user to freeswitch. When trying to add user with user id has different number of digits than 4, it can register but it cant call or cant be called. I changed local extension regular expression in dialpan default.xml but the changes only work with four digits users(user ids or dial number) .So I want dial a number that has five or more digits. What other configurations I must change. Thank you... Burak, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130511/1380d4df/attachment-0001.html From steveayre at gmail.com Sun May 12 02:58:37 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 11 May 2013 23:58:37 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> Message-ID: Run 'netstat -anp | grep freeswitch' and see whether FS is still listening on those ports. Also try restarting sofia / freeswitch to see if that helps. Cal's 'grep LISTEN' command will only show the TCP ports, if SIP arrives on UDP those ports will appear with the above and it'll show all ports (that'll include RTP ports for active calls, but given the problem I'm guessing there won't be any/many). If restarting FreeSWITCH doesn't help and only a reboot seems to help, it would match a bug I have seen in iptables before. Try unloading the SIP conntrack module and allow all packets into your server (ie disable the firewall) and see if they start appearing. I have seen that module randomly start blocking all packets from getting through the firewall but never found why (moving to stateless and avoiding the conntrack module stopped it). That was on an old 2.6 kernel so hopefully isn't present in newer kernels. If just restarting freeswitch helps that's not iptables. Trying to reproduce it on the very latest version of FreeSWITCH would be a good start. -Steve On 11 May 2013 09:24, Christopher Hall wrote: > Steve,**** > > ** ** > > Thanks for your response, I have tried with global siptrace turned on and > I see absolutely nothing when I make the call, the SIP INVITE messages are > sent to the FreeSWITCH host machine as seen using WireShark running on that > host.**** > > ** ** > > Chris.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 10 May 2013 20:01 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE*** > * > > ** ** > > Try enabling the sip trace when it happens to see if FS receives the > packets:**** > > sofia global siptrace on**** > > ** ** > > Steve**** > > ** ** > > ** ** > > > On 10 May 2013, at 17:03, "Christopher Hall" > wrote:**** > > Hi,**** > > **** > > I?m new to FreeSWITCH lists and I?m not sure if this is the correct list > for my post, any help will be much appreciated.**** > > **** > > I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) > SIP service. This works correctly and I can dial in and out using the > service. **** > > **** > > I have an application running on a Windows Server that interfaces using > event socket to the FreeSWITCH server. The intended use is that calls > dialling in through the draytel service are automatically handled by this > application. Again this works correctly.**** > > **** > > However, after a period of time which does not yet appear to be > deterministic the FreeSWITCH server stops responding to INVITE requests > from the draytel server. The FreeSWITCH server can still respond as I can > dial using a local network VOIP phone and the application handles that > call, but on the draytel connection there is no response.**** > > **** > > When FreeSWITCH stops responding I see absolutely nothing in the output > logging information. I have run WireShark on the server that is running > FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the > FreeSWITCH host but I don?t see any kind of response.**** > > **** > > Help!**** > > **** > > Thanks**** > > **** > > Chris.**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130511/20a85a4e/attachment.html From steveayre at gmail.com Sun May 12 03:02:54 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 12 May 2013 00:02:54 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: <017e01ce4e5c$a56a06b0$f03e1410$@vividapps.co.uk> References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> <017e01ce4e5c$a56a06b0$f03e1410$@vividapps.co.uk> Message-ID: Ah, Windows. Ok ignore the last email comments. Well, apart from the latest version. Check the listening ports are still there. The same command will work if you install http://gnuwin32.sourceforge.net/packages/grep.htm - otherwise use 'netstat -anp' and scroll manually. Does FS continue responding to SIP from your softphone? To the port talking to draytel too, even if that's the one you wouldn't normally send to? -Steve On 11 May 2013 16:31, Christopher Hall wrote: > Cal,**** > > ** ** > > Thanks for your response. **** > > ** ** > > I will perform as many of those tests as I can, but while I?m doing that I > need to clarify my situation. On re-reading my original post I realise I > have not been clear.**** > > ** ** > > **? **My FS is running on Windows as a service.**** > > **? **FS is running on the same virtual host as the automated > call handling application.**** > > **? **FS does not stop responding completely but only to SIP > INVITE from the draytel service**** > > **? **When FS is in this state I can still call the automated > call handling application using a soft phone on my LAN.**** > > **? **When FS is in this state I do not see any log output from > FS when the draytel service sends a SIP INVITE.**** > > ** ** > > I hope that is more clear, please let me know if any of the above alters > your thinking. **** > > ** ** > > I will attempt to work through your suggestions.**** > > ** ** > > Chris.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal Leeming > [Simplicity Media Ltd] > *Sent:* 11 May 2013 12:58 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE*** > * > > ** ** > > Hi Chris,**** > > ** ** > > There is a couple of things you can do here to determine what is happening. > **** > > ** ** > > First, make sure you are running the latest freeswitch stable release. > Enable console logging and set the log level to debug (console loglevel > debug) iirc, Next, see if you are able to type 'status' into the CLI after > it has stopped responding, and make sure it outputs something. Perform a > test to see how long it takes for the freeswitch instance to stop > responding. Copy all the log entries from the point where it started, to > the point where it stopped responding, then attach and send the file to > this list. **** > > ** ** > > Once the fs instance has stopped responding, can you please try and > construct a SIP packet and send it manually to the server from the local > host of the server. You can use sendip to do this, by following the > instructions at [1]. Once you send the packet, check to see if it is > received by fs. **** > > ** ** > > If no packet is being received even when sending from localhost, run > "netstat -inp | grep LISTEN | SIPPORTHERE", replacing SIPPORTHERE with the > port you are running sip on fs. Ensure that the only entry for udp is th fs > instance, and nothing else. For a slightly more deeper approach, attach an > strafe to the fs instance, you can do this by looking in the process list > for the fs instance, then typing "strafe -ifq -p PIDHERE" replacing PIDHERE > with the pid of your fs instance. Attempt to send a call to fs, ensure the > packet arrives through wireshark, and see if the packet appears in the > strace (this is a really brutal way of debugging sometimes, it doesn't > offer all the answers but can give you a bit of an X-ray as to what is > going on).**** > > ** ** > > Also ensure you eliminate any possible outside interference, such as > switch gear or routing equipment.. If possible, connect the windows server > and the fs instance directly together, bypassing anything in between. > Ensure you also disable any local firewall on the fs server, I.e. iptables. > **** > > ** ** > > This should hopefully give you a nudge in the right direction to determine > where this problem is happening.**** > > ** ** > > Hope this helps!**** > > ** ** > > Cal**** > > ** ** > > ps) typed on my ipad, sorry for any spelling mistakes**** > > ** ** > > [1] > http://www.moythreads.com/wordpress/2012/03/15/sending-udp-packets-from-the-command-line/ > **** > > ** ** > > ** ** > > > On Saturday, May 11, 2013, Christopher Hall wrote:**** > > Steve,**** > > **** > > Thanks for your response, I have tried with global siptrace turned on and > I see absolutely nothing when I make the call, the SIP INVITE messages are > sent to the FreeSWITCH host machine as seen using WireShark running on that > host.**** > > **** > > Chris.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre > *Sent:* 10 May 2013 20:01 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE*** > * > > **** > > Try enabling the sip trace when it happens to see if FS receives the > packets:**** > > sofia global siptrace on**** > > **** > > Steve**** > > **** > > **** > > > On 10 May 2013, at 17:03, "Christopher Hall" > wrote:**** > > Hi,**** > > **** > > I?m new to FreeSWITCH lists and I?m not sure if this is the correct list > for my post, any help will be much appreciated.**** > > **** > > I have a FreeSWITCH server configured to connect to a Draytel (draytel.org) > SIP service. This works correctly and I can dial in and out using the > service. **** > > **** > > I have an application running on a Windows Server that interfaces using > event socket to the FreeSWITCH server. The intended use is that calls > dialling in through the draytel service are automatically handled by this > application. Again this works correctly.**** > > **** > > However, after a period of time which does not yet appear to be > deterministic the FreeSWITCH server stops responding to INVITE requests > from the draytel server. The FreeSWITCH server can still respond as I can > dial using a local network VOIP phone and the application handles that > call, but on the draytel connection there is no response.**** > > **** > > When FreeSWITCH stops responding I see absolutely nothing in the output > logging information. I have run WireShark on the server that is running > FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the > FreeSWITCH host but I don?t see any kind of response.**** > > **** > > Help!**** > > **** > > Thanks**** > > **** > > Chris.**** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130512/0aceed98/attachment-0001.html From steveayre at gmail.com Sun May 12 03:11:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 12 May 2013 00:11:10 +0100 Subject: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE In-Reply-To: References: <009501ce4d97$eefcd2f0$ccf678d0$@vividapps.co.uk> <773AD15B-66EE-47CB-8E85-F9560E9DB8B2@gmail.com> <011301ce4e21$0489f380$0d9dda80$@vividapps.co.uk> <017e01ce4e5c$a56a06b0$f03e1410$@vividapps.co.uk> Message-ID: Correction, on Windows the netstat options are '-ano' not '-anp' -Steve On 12 May 2013 00:02, Steven Ayre wrote: > Ah, Windows. Ok ignore the last email comments. Well, apart from the > latest version. > > Check the listening ports are still there. The same command will work if > you install http://gnuwin32.sourceforge.net/packages/grep.htm - otherwise > use 'netstat -anp' and scroll manually. > > Does FS continue responding to SIP from your softphone? To the port > talking to draytel too, even if that's the one you wouldn't normally send > to? > > -Steve > > > > On 11 May 2013 16:31, Christopher Hall wrote: > >> Cal,**** >> >> ** ** >> >> Thanks for your response. **** >> >> ** ** >> >> I will perform as many of those tests as I can, but while I?m doing that >> I need to clarify my situation. On re-reading my original post I realise I >> have not been clear.**** >> >> ** ** >> >> **? **My FS is running on Windows as a service.**** >> >> **? **FS is running on the same virtual host as the automated >> call handling application.**** >> >> **? **FS does not stop responding completely but only to SIP >> INVITE from the draytel service**** >> >> **? **When FS is in this state I can still call the automated >> call handling application using a soft phone on my LAN.**** >> >> **? **When FS is in this state I do not see any log output from >> FS when the draytel service sends a SIP INVITE.**** >> >> ** ** >> >> I hope that is more clear, please let me know if any of the above alters >> your thinking. **** >> >> ** ** >> >> I will attempt to work through your suggestions.**** >> >> ** ** >> >> Chris.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Cal >> Leeming [Simplicity Media Ltd] >> *Sent:* 11 May 2013 12:58 >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE** >> ** >> >> ** ** >> >> Hi Chris,**** >> >> ** ** >> >> There is a couple of things you can do here to determine what is >> happening.**** >> >> ** ** >> >> First, make sure you are running the latest freeswitch stable release. >> Enable console logging and set the log level to debug (console loglevel >> debug) iirc, Next, see if you are able to type 'status' into the CLI after >> it has stopped responding, and make sure it outputs something. Perform a >> test to see how long it takes for the freeswitch instance to stop >> responding. Copy all the log entries from the point where it started, to >> the point where it stopped responding, then attach and send the file to >> this list. **** >> >> ** ** >> >> Once the fs instance has stopped responding, can you please try and >> construct a SIP packet and send it manually to the server from the local >> host of the server. You can use sendip to do this, by following the >> instructions at [1]. Once you send the packet, check to see if it is >> received by fs. **** >> >> ** ** >> >> If no packet is being received even when sending from localhost, run >> "netstat -inp | grep LISTEN | SIPPORTHERE", replacing SIPPORTHERE with the >> port you are running sip on fs. Ensure that the only entry for udp is th fs >> instance, and nothing else. For a slightly more deeper approach, attach an >> strafe to the fs instance, you can do this by looking in the process list >> for the fs instance, then typing "strafe -ifq -p PIDHERE" replacing PIDHERE >> with the pid of your fs instance. Attempt to send a call to fs, ensure the >> packet arrives through wireshark, and see if the packet appears in the >> strace (this is a really brutal way of debugging sometimes, it doesn't >> offer all the answers but can give you a bit of an X-ray as to what is >> going on).**** >> >> ** ** >> >> Also ensure you eliminate any possible outside interference, such as >> switch gear or routing equipment.. If possible, connect the windows server >> and the fs instance directly together, bypassing anything in between. >> Ensure you also disable any local firewall on the fs server, I.e. iptables. >> **** >> >> ** ** >> >> This should hopefully give you a nudge in the right direction to >> determine where this problem is happening.**** >> >> ** ** >> >> Hope this helps!**** >> >> ** ** >> >> Cal**** >> >> ** ** >> >> ps) typed on my ipad, sorry for any spelling mistakes**** >> >> ** ** >> >> [1] >> http://www.moythreads.com/wordpress/2012/03/15/sending-udp-packets-from-the-command-line/ >> **** >> >> ** ** >> >> ** ** >> >> >> On Saturday, May 11, 2013, Christopher Hall wrote:**** >> >> Steve,**** >> >> **** >> >> Thanks for your response, I have tried with global siptrace turned on and >> I see absolutely nothing when I make the call, the SIP INVITE messages are >> sent to the FreeSWITCH host machine as seen using WireShark running on that >> host.**** >> >> **** >> >> Chris.**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Steven Ayre >> *Sent:* 10 May 2013 20:01 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] FreeSWITCH not replying to SIP INVITE** >> ** >> >> **** >> >> Try enabling the sip trace when it happens to see if FS receives the >> packets:**** >> >> sofia global siptrace on**** >> >> **** >> >> Steve**** >> >> **** >> >> **** >> >> >> On 10 May 2013, at 17:03, "Christopher Hall" >> wrote:**** >> >> Hi,**** >> >> **** >> >> I?m new to FreeSWITCH lists and I?m not sure if this is the correct list >> for my post, any help will be much appreciated.**** >> >> **** >> >> I have a FreeSWITCH server configured to connect to a Draytel ( >> draytel.org) SIP service. This works correctly and I can dial in and out >> using the service. **** >> >> **** >> >> I have an application running on a Windows Server that interfaces using >> event socket to the FreeSWITCH server. The intended use is that calls >> dialling in through the draytel service are automatically handled by this >> application. Again this works correctly.**** >> >> **** >> >> However, after a period of time which does not yet appear to be >> deterministic the FreeSWITCH server stops responding to INVITE requests >> from the draytel server. The FreeSWITCH server can still respond as I can >> dial using a local network VOIP phone and the application handles that >> call, but on the draytel connection there is no response.**** >> >> **** >> >> When FreeSWITCH stops responding I see absolutely nothing in the output >> logging information. I have run WireShark on the server that is running >> FreeSWITCH and I can see SIP INVITE requests being sent from draytel to the >> FreeSWITCH host but I don?t see any kind of response.**** >> >> **** >> >> Help!**** >> >> **** >> >> Thanks**** >> >> **** >> >> Chris.**** >> >> **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130512/00291e4a/attachment.html From bdfoster at davri.com Sun May 12 19:21:05 2013 From: bdfoster at davri.com (Brian Foster) Date: Sun, 12 May 2013 11:21:05 -0400 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: <15f401ce4e5c$dae01780$90a04680$@bizfocused.com> References: <15f401ce4e5c$dae01780$90a04680$@bizfocused.com> Message-ID: This is exactly why we use the user_exists call. No matter what the user's id is it works. I don't know why that isn't in the default dialplan but if you need an example I can give you one when I get home. -BDF On May 11, 2013 7:07 PM, "Sean Devoy" wrote: > I cannot tell you why this is only a problem with extensions w/ 5 digits, > but I can tell you what failed here. Your underlying error is:**** > > 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, > refer to rfc3711**** > > 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION > ERROR. SDP: > **** > > v=0 > **** > > o=- 3577252345 3577252345 IN IP4 > 141.196.174.60 > **** > > s=pjmedia > **** > > c=IN IP4 > 141.196.174.60 > **** > > t=0 0 > > **** > > m=audio 4010 RTP/AVP 8 0 3 > 101 > **** > > c=IN IP4 > 141.196.174.60 > **** > > a=rtpmap:8 > PCMA/8000 > **** > > a=rtpmap:0 > PCMU/8000 > > **** > > a=rtpmap:3 > GSM/8000 > **** > > a=rtpmap:101 > telephone-event/8000 > **** > > a=fmtp:101 > 0-15 > **** > > a=rtcp:4011 IN IP4 > 192.168.43.10 > > **** > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik > **** > > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO > **** > > ** ** > > 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK]**** > > 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 > (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> > HANGUP **** > > 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup > sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION]**** > > ** ** > > Can you try it without the crypto stuff and paste the same output to > pastebin.freeswitch.org (not here in email)?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Burak > BorYazilim > *Sent:* Saturday, May 11, 2013 7:28 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch User Adding**** > > ** ** > > First off all thanks for your kind helps.**** > > ** ** > > Let me tell you the situation more clearly.**** > > ** ** > > Firstly I used Android phones as sip clients with the open source sip > client program, CSipSimple. My server computer has Ubuntu 12.04 LTS with > one static ip address.**** > > freeswitch version output: FreeSWITCH Version > 1.3.13b+git~20130205T003128Z~70a9560306 (git 70a9560 2013-02-05 00:31:28Z) > **** > > We were using the system with 4 digits number sip account names without > any problem. Also the account name will be used as dial number. An example > of the user xml file is below. I changed the regex in dialpan/default.xml. > The changings were perfectly succesfull with, again, 4 digits number. But > when trying to include any other number of digits (3, 5 and 6 were tested), > it doesn't work(When I change the regex to accept only 5 digits numbers, 4 > digits ones didnt work as expected). I really could not understand why it > is happening. Why there was no problem with 4 digits number and why the > exact same system does not work with this basic change.**** > > ** ** > > To explain the errors more, I want to talk about my tests. Firstly I used > 4 digits number user. This test repeated with tls and srtp. These two test > were succesfull. **** > > And the secand test is same with first test but with 5 digits numbers > without tls and srtp. Registration was succesfull but cant call. (or the > call could not be forwarded) Of course I changed the regex and execute > reloadxml in this test. GSM (8kHz) and G722(16kHz) codecs were used. As you > see below, CODEC NEGOTIATION ERROR occured. After getting this error I > changed the codes as SILK(16kHz). With this change I made the third test. > Third test result is also below. **** > > ** ** > > show registretions output: **** > > > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata > **** > > > 60022,my.server.ip.address,oOnxiVJFAgQWLABrD01UsYTVOY3TVSlx,sofia/internal/sip:60022 at 141.196.174.60:52245 > ;ob,1368271435,141.196.174.60,52245,udp,server,**** > > > 60021,my.server.ip.address,5r4MTDiZPhs5qzdin9A3hEUh1zZsdqqk,sofia/internal/sip:60021 at 141.196.174.60:57938 > ;ob,1368271446,141.196.174.60,57938,udp,server,**** > > ** ** > > ** ** > > @Sean, sorry but I cant post my whole log file because my server has got a > network attack so I am quite busy with this attacker. And also I dont think > CSipSimple has a problem with 5 digits because this system (with 5 digits) > was succesfull with Kamailio and CSipSimple.**** > > ** ** > > If I could not be clear please let me know. Thanks...**** > > ** ** > > user xml example: This was exact same with the default users in > freeswitch. Only the number changed.**** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** > ** > > **** > > **** > > * > *** > > **** > > value="$${outbound_caller_name}"/>**** > > value="$${outbound_caller_id}"/>**** > > **** > > **** > > **** > > **** > > ** ** > > ** ** > > ** ** > > The second test error:**** > > ** ** > > 2013-05-11 12:13:56.800443 [DEBUG] sofia_reg.c:1511 Send challenge for > [60021 at my.server.ip.address]**** > > 2013-05-11 12:13:56.900444 [DEBUG] sofia_reg.c:1511 Send challenge for > [60021 at my.server.ip.address] > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'record_stereo' = 'true' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_gateway' = 'example.com' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_areacode' = '918' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'transfer_fallback_extension' = 'operator' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'toll_allow' = 'domestic,international,local' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'accountcode' = '60021' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'user_context' = 'default' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_name' = 'Extension 60021' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_number' = '60021' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_name' = 'FreeSWITCH' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_number' = '0000000000' > **** > > 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'callgroup' = 'techsupport' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'record_stereo' = 'true' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_gateway' = 'example.com' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_areacode' = '918' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'transfer_fallback_extension' = 'operator' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'toll_allow' = 'domestic,international,local' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'accountcode' = '60021' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'user_context' = 'default' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_name' = 'Extension 60021' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_number' = '60021' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_name' = 'FreeSWITCH' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_number' = '0000000000' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'callgroup' = 'techsupport' > **** > > 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:1683 Register: > > **** > > From: [60021 at my.server.ip.address] > > **** > > Contact: ["user" ] > > **** > > Expires: [900] > > **** > > 2013-05-11 12:14:06.820443 [DEBUG] sofia_reg.c:1511 Send challenge for > [60022 at my.server.ip.address] > **** > > 2013-05-11 12:14:06.900444 [DEBUG] sofia_reg.c:1511 Send challenge for > [60022 at my.server.ip.address] > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'record_stereo' = 'true' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_gateway' = 'example.com' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_areacode' = '918' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'transfer_fallback_extension' = 'operator' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'toll_allow' = 'domestic,international,local' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'accountcode' = '60022' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'user_context' = 'default' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_name' = 'Extension 60022' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_number' = '60022' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_name' = 'FreeSWITCH' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_number' = '0000000000' > **** > > 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'callgroup' = 'techsupport' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'record_stereo' = 'true' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_gateway' = 'example.com' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_areacode' = '918' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'transfer_fallback_extension' = 'operator' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'toll_allow' = 'domestic,international,local' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'accountcode' = '60022' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'user_context' = 'default' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_name' = 'Extension 60022' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_number' = '60022' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_name' = 'FreeSWITCH' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_number' = '0000000000' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'callgroup' = 'techsupport' > **** > > 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:1683 Register: > > **** > > From: [60022 at my.server.ip.address] > > **** > > Contact: ["user" ] > > **** > > Expires: [900] > > **** > > 2013-05-11 12:14:14.640437 [NOTICE] switch_channel.c:968 New Channel > sofia/internal/60021 at my.server.ip.address[26730cd2-ba1b-11e2-acc5-bda7cbfd9554] > **** > > 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK]**** > > 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/60021 at my.server.ip.address) Running State Change CS_NEW > **** > > 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:433 ( > sofia/internal/60021 at my.server.ip.address) State NEW > **** > > 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected > by acl "domains". Falling back to Digest auth. > **** > > 2013-05-11 12:14:14.660438 [DEBUG] sofia_reg.c:1511 Send challenge for > [60022 at my.server.ip.address] > **** > > 2013-05-11 12:14:14.660438 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:1719 detaching session > 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 > **** > > 2013-05-11 12:14:14.780439 [DEBUG] sofia.c:1811 Re-attaching to session > 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 > **** > > 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:7733 IP 141.196.174.60 Rejected > by acl "domains". Falling back to Digest auth. > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'record_stereo' = 'true' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_gateway' = 'example.com' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'default_areacode' = '918' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'transfer_fallback_extension' = 'operator' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'toll_allow' = 'domestic,international,local' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'accountcode' = '60021' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'user_context' = 'default' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_name' = 'Extension 60021' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'effective_caller_id_number' = '60021' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_name' = 'FreeSWITCH' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'outbound_caller_id_number' = '0000000000' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> > 'callgroup' = 'techsupport' > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel > sofia/internal/60021 at my.server.ip.address entering state [received][100] > **** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5589 Remote SDP: > > **** > > v=0 > > **** > > o=- 3577252345 3577252345 IN IP4 141.196.174.60 > > **** > > s=pjmedia > > **** > > c=IN IP4 141.196.174.60 > > **** > > t=0 0 > > **** > > m=audio 4010 RTP/AVP 8 0 3 101 > > **** > > c=IN IP4 141.196.174.60 > > **** > > a=rtpmap:8 PCMA/8000 > > **** > > a=rtpmap:0 PCMU/8000 > > **** > > a=rtpmap:3 GSM/8000 > > **** > > a=rtpmap:101 telephone-event/8000 > > **** > > a=fmtp:101 0-15 > > **** > > a=rtcp:4011 IN IP4 192.168.43.10 > > **** > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik > **** > > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO > **** > > ** ** > > 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5802 ( > sofia/internal/60021 at my.server.ip.address) State Change CS_NEW -> CS_INIT > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/60021 at my.server.ip.address) Running State Change CS_INIT > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( > sofia/internal/60021 at my.server.ip.address) State INIT > **** > > 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 > sofia/internal/60021 at my.server.ip.address SOFIA INIT > **** > > 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 ( > sofia/internal/60021 at my.server.ip.address) State Change CS_INIT -> > CS_ROUTING **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal > sofia/internal/60021 at my.server.ip.address [BREAK] > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( > sofia/internal/60021 at my.server.ip.address) State INIT going to sleep > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( > sofia/internal/60021 at my.server.ip.address) Running State Change > CS_ROUTING **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 ( > sofia/internal/60021 at my.server.ip.address) Callstate Change DOWN -> > RINGING **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 ( > sofia/internal/60021 at my.server.ip.address) State ROUTING > **** > > 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 > sofia/internal/60021 at my.server.ip.address SOFIA ROUTING > **** > > 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:117 > sofia/internal/60021 at my.server.ip.address Standard ROUTING > **** > > 2013-05-11 12:14:14.800439 [INFO] mod_dialplan_xml.c:557 Processing 60021 > <60021>->60022 in context default**** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->unloop] continue=false**** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->tod_example] continue=true > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) > [tod_example] break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->holiday_example] continue=true > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch (FAIL) > [holiday_example] break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->global-intercept] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [global-intercept] destination_number(60022) =~ /^886$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->group-intercept] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [group-intercept] destination_number(60022) =~ /^\*8$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->intercept-ext] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [intercept-ext] destination_number(60022) =~ /^\*\*(\d+)$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->redial] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [redial] > destination_number(60022) =~ /^(redial|870)$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->global] continue=true > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] > ${call_debug}(false) =~ /^true$/ break=never > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) [global] > ${sip_has_crypto}() =~ > /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] > ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED NEGOTIATION)/ > break=on-false **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) [global] > ${switch_r_sdp}(v=0 > **** > > o=- 3577252345 3577252345 IN IP4 141.196.174.60 > > **** > > s=pjmedia > > **** > > c=IN IP4 141.196.174.60 > > **** > > t=0 0 > > **** > > m=audio 4010 RTP/AVP 8 0 3 101 > > **** > > c=IN IP4 141.196.174.60 > > **** > > a=rtpmap:8 PCMA/8000 > > **** > > a=rtpmap:0 PCMU/8000 > > **** > > a=rtpmap:3 GSM/8000 > > **** > > a=rtpmap:101 telephone-event/8000 > > **** > > a=fmtp:101 0-15 > > **** > > a=rtcp:4011 IN IP4 192.168.43.10 > > **** > > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik > **** > > a=crypto:2 AES_CM_128_HMAC_SHA1_32 > inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO > **** > > ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never > > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(sip_secure_media=true) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > export(sip_secure_media=true) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Absolute Condition > [global] > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-last_dial/global/${uuid}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->snom-demo-2] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [snom-demo-2] destination_number(60022) =~ /^9001$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->snom-demo-1] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [snom-demo-1] destination_number(60022) =~ /^9000$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->eavesdrop] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [eavesdrop] destination_number(60022) =~ /^88(\d{4})$|^\*0(.*)$/ > break=on-false **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->eavesdrop] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [eavesdrop] destination_number(60022) =~ /^779$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->call_return] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [call_return] destination_number(60022) =~ /^\*69$|^869$|^lcr$/ > break=on-false **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->del-group] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [del-group] destination_number(60022) =~ /^80(\d{2})$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->add-group] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [add-group] destination_number(60022) =~ /^81(\d{2})$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->call-group-simo] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [call-group-simo] destination_number(60022) =~ /^82(\d{2})$/ break=on-false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->call-group-order] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [call-group-order] destination_number(60022) =~ /^83(\d{2})$/ > break=on-false **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->extension-intercom] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) > [extension-intercom] destination_number(60022) =~ /^8(10[01][0-9])$/ > break=on-false **** > > Dialplan: sofia/internal/60021 at my.server.ip.address parsing > [default->Local_Extension] continue=false > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) > [Local_Extension] destination_number(60022) =~ > /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false > > > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > export(dialed_extension=60022) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > bind_meta_app(1 b s execute_extension::dx XML features) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > bind_meta_app(2 b s > record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) > > > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > bind_meta_app(3 b s execute_extension::cf XML features) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > bind_meta_app(4 b s execute_extension::att_xfer XML features) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(ringback=${us-ring}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(transfer_ringback=local_stream://moh) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(call_timeout=30) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(hangup_after_bridge=true) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(continue_on_fail=true) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} > var callgroup)}) **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) > **** > > Dialplan: sofia/internal/60021 at my.server.ip.address Action > hash(insert/${domain_name}-last_dial_ext/global/${uuid}) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130512/765b3bb6/attachment-0001.html From spencer at 5ninesolutions.com Mon May 13 01:54:42 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Sun, 12 May 2013 17:54:42 -0400 Subject: [Freeswitch-users] TCP ACK Ping Message-ID: <51900FA2.9020509@5ninesolutions.com> Hello all, I have been doing some experimentation migrating several phones from UDP to TCP SIP to address MTU issues with BLF NOTIFYs. Mostly everything is working great except for Polycom handsets running UC v4. (Note that I did not test any v3 phones.) The phones are behind NAT and Freeswitch is on a public IP. The phones periodically send a TCP ACK ping to Freeswitch which goes unanswered and then phone then tears down the TCP connection. I'm not at all a TCP expert but should Freeswitch be responding to this unsolicited ACK? I have tried in vain to disable these keepalives from the Polycoms. Has anyone else encountered this behavior? Thanks in advance, Spencer From steveayre at gmail.com Mon May 13 02:37:25 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 12 May 2013 23:37:25 +0100 Subject: [Freeswitch-users] NO_ROUTE_DESTINATION In-Reply-To: <1368251030474-7590575.post@n2.nabble.com> References: <1368008894683-7590466.post@n2.nabble.com> <1368251030474-7590575.post@n2.nabble.com> Message-ID: Not sure why you replaced the gateway name with xxxx - it makes it impossible to see which gateway you used without looking through the log. It isn't hiding anything of value. As I said on my previous email, but in more detail: EXECUTE sofia/internal/407 at 10.15.1.41bridge(sofia/gateway/globe/0018444485452) You are dialing out via gateway 'globe' external::globe gateway sip:029080340 at siphosted.com > REGED The 'globe' gateway is configured on the 'external' profile. Any call sent OUT via the globe gateway will leave from the profile it is configured on. If your call comes IN via internal, it will go OUT via external since the gateway is on that profile. FS lets you dial between profiles (and even different protocols) so what you came in on does not affect what you leave on. This would be the expected and indeed required behaviour, since the gateway configured on external uses its IP and the siphosted.com provider may not be reachable from the IP used on other profiles. -Steve On 11 May 2013 06:43, baskar wrote: > Hi Steve, > > Thanks for the reply. > > When i try to use only the gateway it pass through external profile why it > is happning. > > SIP trunk has username and password with hard cable connected from service > provider. > > SIP trunk configured in freeswitch server and pass through service provider > through SIP trunk. > > sofia/internal/407 at 10.15.1.41 bridge(sofia/gateway/xxxx/0018444485452) > 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel > sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f] > > Can any one guide me procedure to calls passing through the SIP server. > > Thanks in advance. > > Regards, > N.Baskar > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.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://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/20130512/b304d6b1/attachment.html From steveayre at gmail.com Mon May 13 02:39:21 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 12 May 2013 23:39:21 +0100 Subject: [Freeswitch-users] NO_ROUTE_DESTINATION In-Reply-To: References: <1368008894683-7590466.post@n2.nabble.com> <1368251030474-7590575.post@n2.nabble.com> Message-ID: 1. recv 369 bytes from udp/[202.126.41.108]:5060 at 22:13:09.398177: 2. ------------------------------------------------------------------------ 3. SIP/2.0 604 Does not exist anywhere The NO_ROUTE_DESTINATION clearing cause is due to the 604 sent to you from your SIP provider. It is unrelated to your profile. Only your SIP provider can tell you the reason for the 604 - raise it with them. It could be a number of things - an invalid number, unreachable number, internal/upstream error etc. -Steve On 12 May 2013 23:37, Steven Ayre wrote: > Not sure why you replaced the gateway name with xxxx - it makes it > impossible to see which gateway you used without looking through the log. > It isn't hiding anything of value. > > As I said on my previous email, but in more detail: > > EXECUTE sofia/internal/407 at 10.15.1.41bridge(sofia/gateway/globe/0018444485452) > > > You are dialing out via gateway 'globe' > > external::globe gateway sip:029080340 at siphosted.com REGED > > > The 'globe' gateway is configured on the 'external' profile. > > Any call sent OUT via the globe gateway will leave from the profile it is > configured on. > > If your call comes IN via internal, it will go OUT via external since the > gateway is on that profile. FS lets you dial between profiles (and even > different protocols) so what you came in on does not affect what you leave > on. > > This would be the expected and indeed required behaviour, since the > gateway configured on external uses its IP and the siphosted.com provider > may not be reachable from the IP used on other profiles. > > -Steve > > > > > On 11 May 2013 06:43, baskar wrote: > >> Hi Steve, >> >> Thanks for the reply. >> >> When i try to use only the gateway it pass through external profile why it >> is happning. >> >> SIP trunk has username and password with hard cable connected from service >> provider. >> >> SIP trunk configured in freeswitch server and pass through service >> provider >> through SIP trunk. >> >> sofia/internal/407 at 10.15.1.41 bridge(sofia/gateway/xxxx/0018444485452) >> 2013-05-08 06:13:09.376846 [NOTICE] switch_channel.c:669 New Channel >> sofia/external/0018444485452 [4cdd9992-b763-11e2-bdef-2d40d364119f] >> >> Can any one guide me procedure to calls passing through the SIP server. >> >> Thanks in advance. >> >> Regards, >> N.Baskar >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/NO-ROUTE-DESTINATION-tp7590466p7590575.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://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/20130512/63af0390/attachment.html From tom at zuit.net Mon May 13 04:00:05 2013 From: tom at zuit.net (Tom Berchenbriter) Date: Mon, 13 May 2013 00:00:05 +0000 Subject: [Freeswitch-users] FW: failed registrations In-Reply-To: References: Message-ID: I have this freeswitch box im trying to get registered to my zoiper sip client, with no success... Its reaching the box and the service, but I cant get it to register. Please help! freeswitch at internal> sofia profile internal siptrace on Enabled sip debugging on internal recv 560 bytes from udp/[76.114.1.252]:47607 at 23:39:05.536725: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at 69.1.72.182:5060 SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-d2becb02f515cd0e-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter";tag=Lbi6IA1V3zXG From: "Tom Berchenbriter";tag=ce466e18 Call-ID: MGZlMTk5YTU1YjE3ODcxOWIwMThlNjc0NzllOGFkMGM. CSeq: 3 SUBSCRIBE Expires: 0 User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ send 858 bytes to udp/[76.114.1.252]:47607 at 23:39:05.538257: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-d2becb02f515cd0e-1---d8754z- From: "Tom Berchenbriter";tag=ce466e18 To: "Tom Berchenbriter" ;tag=Lbi6IA1V3zXG Call-ID: MGZlMTk5YTU1YjE3ODcxOWIwMThlNjc0NzllOGFkMGM. CSeq: 3 SUBSCRIBE Contact: Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 ------------------------------------------------------------------------ recv 744 bytes from udp/[76.114.1.252]:47607 at 23:39:05.724616: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-532643e8d1a83a50-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=d8034e7b Call-ID: YjRjZTBhNGRjZmE4NTZkMjYzM2UxYzcxZjMzNzVlOTY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:39:05.718477 [WARNING] sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 751 bytes to udp/[76.114.1.252]:47607 at 23:39:05.726015: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-532643e8d1a83a50-1---d8754z- From: "Tom Berchenbriter";tag=d8034e7b To: "Tom Berchenbriter" ;tag=mHNr3aaUjDjmB Call-ID: YjRjZTBhNGRjZmE4NTZkMjYzM2UxYzcxZjMzNzVlOTY. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="freeswitch.vunity.com", nonce="225a9c30-bb5d-11e2-861f-c7f05fc7f0f9", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 791 bytes from udp/[76.114.1.252]:47607 at 23:39:05.762650: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-3e1fa89fd20d46a9-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=a33d7255 Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 1 SUBSCRIBE Expires: 3600 Accept: application/simple-message-summary Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 852 bytes to udp/[76.114.1.252]:47607 at 23:39:05.764152: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-3e1fa89fd20d46a9-1---d8754z- From: "Tom Berchenbriter";tag=a33d7255 To: "Tom Berchenbriter" ;tag=urWr0mjMfIzM Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 1 SUBSCRIBE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ send 1009 bytes to udp/[76.114.1.252]:47607 at 23:39:05.782511: ------------------------------------------------------------------------ NOTIFY sip:tom at 76.114.1.252:47607;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 69.1.72.182;rport;branch=z9hG4bK7tvyyD97yUv0S Max-Forwards: 70 From: "Tom Berchenbriter" ;tag=urWr0mjMfIzM To: "Tom Berchenbriter" ;tag=a33d7255 Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 551189062 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/simple-message-summary Content-Length: 62 Messages-Waiting: no Message-Account: sip:tom at 69.1.72.182 ------------------------------------------------------------------------ recv 1024 bytes from udp/[76.114.1.252]:47607 at 23:39:05.804518: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-52fbb637013f5606-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=d8034e7b Call-ID: YjRjZTBhNGRjZmE4NTZkMjYzM2UxYzcxZjMzNzVlOTY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Authorization: Digest username="tom",realm="freeswitch.vunity.com",nonce="225a9c30-bb5d-11e2-861f-c7f05fc7f0f9",uri="sip:freeswitch.vunity.com;transport=UDP",response="d3c6288692ba20aec6cfd6618d94e1fb",cnonce="3f10e3fd1737677bc297b6e32182d802",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:39:05.798483 [WARNING] sofia_reg.c:1390 SIP auth failure (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 619 bytes to udp/[76.114.1.252]:47607 at 23:39:05.805865: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-52fbb637013f5606-1---d8754z- From: "Tom Berchenbriter";tag=d8034e7b To: "Tom Berchenbriter" ;tag=p37960B2cZySj Call-ID: YjRjZTBhNGRjZmE4NTZkMjYzM2UxYzcxZjMzNzVlOTY. CSeq: 2 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 464 bytes from udp/[76.114.1.252]:47607 at 23:39:05.868464: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.72.182;rport=5060;branch=z9hG4bK7tvyyD97yUv0S Contact: To: "Tom Berchenbriter";tag=a33d7255 From: "Tom Berchenbriter";tag=urWr0mjMfIzM Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 551189062 NOTIFY User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Content-Length: 0 ------------------------------------------------------------------------ recv 560 bytes from udp/[76.114.1.252]:47607 at 23:51:33.620568: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at 69.1.72.182:5060 SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-7ddfdee4a1e63c23-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter";tag=urWr0mjMfIzM From: "Tom Berchenbriter";tag=a33d7255 Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 3 SUBSCRIBE Expires: 0 User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ send 858 bytes to udp/[76.114.1.252]:47607 at 23:51:33.622141: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-7ddfdee4a1e63c23-1---d8754z- From: "Tom Berchenbriter";tag=a33d7255 To: "Tom Berchenbriter" ;tag=urWr0mjMfIzM Call-ID: NmIyNWNmNWExY2E0NGU4ZjkyNzkyNDUxMjhjZmQ4ZTM. CSeq: 3 SUBSCRIBE Contact: Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 ------------------------------------------------------------------------ recv 791 bytes from udp/[76.114.1.252]:47607 at 23:51:33.794491: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-70517ab3d3d87d57-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=975cc531 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 1 SUBSCRIBE Expires: 3600 Accept: application/simple-message-summary Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 852 bytes to udp/[76.114.1.252]:47607 at 23:51:33.796922: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-70517ab3d3d87d57-1---d8754z- From: "Tom Berchenbriter";tag=975cc531 To: "Tom Berchenbriter" ;tag=Nya4pme1sMr4 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 1 SUBSCRIBE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ recv 744 bytes from udp/[76.114.1.252]:47607 at 23:51:33.837064: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-3d4439830a743f73-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=14095d76 Call-ID: ZmM5ZTI5OGNhOWU0MzRiMzNhM2M3ZWU3YTNkYWRlODY. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:51:33.818490 [WARNING] sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 751 bytes to udp/[76.114.1.252]:47607 at 23:51:33.838301: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-3d4439830a743f73-1---d8754z- From: "Tom Berchenbriter";tag=14095d76 To: "Tom Berchenbriter" ;tag=rNtUaQD96gBZS Call-ID: ZmM5ZTI5OGNhOWU0MzRiMzNhM2M3ZWU3YTNkYWRlODY. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="freeswitch.vunity.com", nonce="e0437c48-bb5e-11e2-8620-c7f05fc7f0f9", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ send 1009 bytes to udp/[76.114.1.252]:47607 at 23:51:33.887280: ------------------------------------------------------------------------ NOTIFY sip:tom at 76.114.1.252:47607;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 69.1.72.182;rport;branch=z9hG4bK83NQ08SBv4jKN Max-Forwards: 70 From: "Tom Berchenbriter" ;tag=Nya4pme1sMr4 To: "Tom Berchenbriter" ;tag=975cc531 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 551189063 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/simple-message-summary Content-Length: 62 Messages-Waiting: no Message-Account: sip:tom at 69.1.72.182 ------------------------------------------------------------------------ recv 464 bytes from udp/[76.114.1.252]:47607 at 23:51:34.020407: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.72.182;rport=5060;branch=z9hG4bK83NQ08SBv4jKN Contact: To: "Tom Berchenbriter";tag=975cc531 From: "Tom Berchenbriter";tag=Nya4pme1sMr4 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 551189063 NOTIFY User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Content-Length: 0 ------------------------------------------------------------------------ recv 1024 bytes from udp/[76.114.1.252]:47607 at 23:51:34.411668: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-8363984eff48ee06-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=14095d76 Call-ID: ZmM5ZTI5OGNhOWU0MzRiMzNhM2M3ZWU3YTNkYWRlODY. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Authorization: Digest username="tom",realm="freeswitch.vunity.com",nonce="e0437c48-bb5e-11e2-8620-c7f05fc7f0f9",uri="sip:freeswitch.vunity.com;transport=UDP",response="cbf32836c98ddeb2ebcba998cf2e56e8",cnonce="b2a92947dd5a5438d1b2bf72461d27a6",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:51:34.398556 [WARNING] sofia_reg.c:1390 SIP auth failure (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 619 bytes to udp/[76.114.1.252]:47607 at 23:51:34.414573: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-8363984eff48ee06-1---d8754z- From: "Tom Berchenbriter";tag=14095d76 To: "Tom Berchenbriter" ;tag=SyKmcjyc4S1HN Call-ID: ZmM5ZTI5OGNhOWU0MzRiMzNhM2M3ZWU3YTNkYWRlODY. CSeq: 2 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 415 bytes from udp/[199.19.105.189]:5062 at 23:51:41.627346: ------------------------------------------------------------------------ OPTIONS sip:100 at 69.1.72.182 SIP/2.0 Via: SIP/2.0/UDP 199.19.105.170:5062;branch=z9hG4bK-4277677944;rport Content-Length: 0 From: "sipvicious"; tag=3435303134386236313363340132383932333932333931 Accept: application/sdp User-Agent: friendly-scanner To: "sipvicious" Contact: sip:100 at 199.19.105.170:5062 CSeq: 1 OPTIONS Call-ID: 172752228514132614325030 Max-Forwards: 70 ------------------------------------------------------------------------ send 765 bytes to udp/[199.19.105.189]:5062 at 23:51:41.628761: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 199.19.105.170:5062;branch=z9hG4bK-4277677944;rport=5062;received=199.19.105.189 From: "sipvicious"; tag=3435303134386236313363340132383932333932333931 To: "sipvicious" ;tag=t7cDeDFg12Q4g Call-ID: 172752228514132614325030 CSeq: 1 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ recv 560 bytes from udp/[76.114.1.252]:47607 at 23:51:45.494622: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at 69.1.72.182:5060 SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-434ebc01bd6f2fac-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter";tag=Nya4pme1sMr4 From: "Tom Berchenbriter";tag=975cc531 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 3 SUBSCRIBE Expires: 0 User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ recv 791 bytes from udp/[76.114.1.252]:47607 at 23:51:45.494928: ------------------------------------------------------------------------ SUBSCRIBE sip:tom at freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-7c668ae5b804369f-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=f00fc03b Call-ID: YzNhOGRkNjg4N2YwYTYxMjFhNGVhODQ4ZGVhNjVlMmU. CSeq: 1 SUBSCRIBE Expires: 3600 Accept: application/simple-message-summary Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Event: message-summary Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ recv 744 bytes from udp/[76.114.1.252]:47607 at 23:51:45.495128: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-0fbb0b4663e0b0be-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=751af136 Call-ID: YjkzODg0MjIxZmY5OGM2ODk2MDJhOGExZmY4YjZhY2E. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ send 858 bytes to udp/[76.114.1.252]:47607 at 23:51:45.497565: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-434ebc01bd6f2fac-1---d8754z- From: "Tom Berchenbriter";tag=975cc531 To: "Tom Berchenbriter" ;tag=Nya4pme1sMr4 Call-ID: Mjg0MzM2YTE3YjJhMTViYjIxZWQ3YzcyMDc5ODZhNzA. CSeq: 3 SUBSCRIBE Contact: Expires: 0 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Length: 0 ------------------------------------------------------------------------ send 852 bytes to udp/[76.114.1.252]:47607 at 23:51:45.497742: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-7c668ae5b804369f-1---d8754z- From: "Tom Berchenbriter";tag=f00fc03b To: "Tom Berchenbriter" ;tag=usl4KsioRij6 Call-ID: YzNhOGRkNjg4N2YwYTYxMjFhNGVhODQ4ZGVhNjVlMmU. CSeq: 1 SUBSCRIBE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:51:45.478474 [WARNING] sofia_reg.c:1445 SIP auth challenge (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 751 bytes to udp/[76.114.1.252]:47607 at 23:51:45.498204: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-0fbb0b4663e0b0be-1---d8754z- From: "Tom Berchenbriter";tag=751af136 To: "Tom Berchenbriter" ;tag=vSZyH3gQUm49Q Call-ID: YjkzODg0MjIxZmY5OGM2ODk2MDJhOGExZmY4YjZhY2E. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="freeswitch.vunity.com", nonce="e736a4da-bb5e-11e2-8621-c7f05fc7f0f9", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 1024 bytes from udp/[76.114.1.252]:47607 at 23:51:45.596784: ------------------------------------------------------------------------ REGISTER sip:freeswitch.vunity.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-01ece8a9c2d93bbc-1---d8754z- Max-Forwards: 70 Contact: To: "Tom Berchenbriter" From: "Tom Berchenbriter";tag=751af136 Call-ID: YjkzODg0MjIxZmY5OGM2ODk2MDJhOGExZmY4YjZhY2E. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Authorization: Digest username="tom",realm="freeswitch.vunity.com",nonce="e736a4da-bb5e-11e2-8621-c7f05fc7f0f9",uri="sip:freeswitch.vunity.com;transport=UDP",response="df5194bf54c6211ea0c123d7fb8c27c0",cnonce="0b7ff76d70148748b0a7876d76253712",nc=00000001,qop=auth,algorithm=MD5 Allow-Events: presence, kpml Content-Length: 0 ------------------------------------------------------------------------ 2013-05-12 16:51:45.578812 [WARNING] sofia_reg.c:1390 SIP auth failure (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 send 619 bytes to udp/[76.114.1.252]:47607 at 23:51:45.598577: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 76.114.1.252:47607;branch=z9hG4bK-d8754z-01ece8a9c2d93bbc-1---d8754z- From: "Tom Berchenbriter";tag=751af136 To: "Tom Berchenbriter" ;tag=X2rQKy1trXtvK Call-ID: YjkzODg0MjIxZmY5OGM2ODk2MDJhOGExZmY4YjZhY2E. CSeq: 2 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 1009 bytes to udp/[76.114.1.252]:47607 at 23:51:45.602129: ------------------------------------------------------------------------ NOTIFY sip:tom at 76.114.1.252:47607;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 69.1.72.182;rport;branch=z9hG4bK9cFg23aFSD95g Max-Forwards: 70 From: "Tom Berchenbriter" ;tag=usl4KsioRij6 To: "Tom Berchenbriter" ;tag=f00fc03b Call-ID: YzNhOGRkNjg4N2YwYTYxMjFhNGVhODQ4ZGVhNjVlMmU. CSeq: 551189064 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-086cbf1 2012-05-03 13-17-05 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/simple-message-summary Content-Length: 62 Messages-Waiting: no Message-Account: sip:tom at 69.1.72.182 ------------------------------------------------------------------------ recv 464 bytes from udp/[76.114.1.252]:47607 at 23:51:45.684311: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 69.1.72.182;rport=5060;branch=z9hG4bK9cFg23aFSD95g Contact: To: "Tom Berchenbriter";tag=f00fc03b From: "Tom Berchenbriter";tag=usl4KsioRij6 Call-ID: YzNhOGRkNjg4N2YwYTYxMjFhNGVhODQ4ZGVhNjVlMmU. CSeq: 551189064 NOTIFY User-Agent: Zoiper Communicator 2.05.11136 rev.11135 Content-Length: 0 ------------------------------------------------------------------------ freeswitch at internal> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/9541bd7a/attachment-0001.html From b2m at a-cti.com Mon May 13 10:17:21 2013 From: b2m at a-cti.com (Bala Murugan Mahendran) Date: Mon, 13 May 2013 11:47:21 +0530 Subject: [Freeswitch-users] Strange CDR entry Message-ID: I have lot of entries like 1000,1001,1002 and so on, Is someone trying to get inside? I believe they didn't in yet but how come we have cdr log like this? "1004","1004","012972598371070","public","2013-05-12 07:30:07","","2013-05-12 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" "1004","1004","013972598371070","public","2013-05-12 07:30:07","","2013-05-12 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" "1004","1004","014972598371070","public","2013-05-12 07:30:07","","2013-05-12 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" "1004","1004","010972598371070","public","2013-05-12 07:30:08","","2013-05-12 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" "1004","1004","0061972598371070","public","2013-05-12 07:30:08","","2013-05-12 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" "1004","1004","0041972598371070","public","2013-05-12 07:30:09","","2013-05-12 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" "1004","1004","000972598371070","public","2013-05-12 07:30:09","","2013-05-12 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" "1004","1004","006972598371070","public","2013-05-12 07:30:09","","2013-05-12 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" "1004","1004","002972598371070","public","2013-05-12 07:30:10","","2013-05-12 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" "1004","1004","810972598371070","public","2013-05-12 07:29:49","","2013-05-12 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/16c5b1b8/attachment.html From mburakbor at gmail.com Mon May 13 10:32:19 2013 From: mburakbor at gmail.com (=?ISO-8859-9?Q?Burak_BorYaz=FDl=FDm?=) Date: Mon, 13 May 2013 09:32:19 +0300 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: <15f401ce4e5c$dae01780$90a04680$@bizfocused.com> Message-ID: @Sean what do you mean with crypto stuff? If you mean tls and srtp, I didnt use them while testing. I am still trying to solve what is changing when the number digit length is changed? @Brian yes an example will be helpful if you have time for it, I will be very pleased. 2013/5/12 Brian Foster > This is exactly why we use the user_exists call. No matter what the user's > id is it works. I don't know why that isn't in the default dialplan but if > you need an example I can give you one when I get home. > > -BDF > On May 11, 2013 7:07 PM, "Sean Devoy" wrote: > >> I cannot tell you why this is only a problem with extensions w/ 5 digits, >> but I can tell you what failed here. Your underlying error is:**** >> >> 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, >> refer to rfc3711**** >> >> 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION >> ERROR. SDP: >> **** >> >> v=0 >> **** >> >> o=- 3577252345 3577252345 IN IP4 >> 141.196.174.60 >> **** >> >> s=pjmedia >> **** >> >> c=IN IP4 >> 141.196.174.60 >> **** >> >> t=0 0 >> >> **** >> >> m=audio 4010 RTP/AVP 8 0 3 >> 101 >> **** >> >> c=IN IP4 >> 141.196.174.60 >> **** >> >> a=rtpmap:8 >> PCMA/8000 >> **** >> >> a=rtpmap:0 >> PCMU/8000 >> >> **** >> >> a=rtpmap:3 >> GSM/8000 >> **** >> >> a=rtpmap:101 >> telephone-event/8000 >> **** >> >> a=fmtp:101 >> 0-15 >> **** >> >> a=rtcp:4011 IN IP4 >> 192.168.43.10 >> >> **** >> >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >> **** >> >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >> **** >> >> ** ** >> >> 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK]**** >> >> 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 >> (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> >> HANGUP **** >> >> 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup >> sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] >> [INCOMPATIBLE_DESTINATION]**** >> >> ** ** >> >> Can you try it without the crypto stuff and paste the same output to >> pastebin.freeswitch.org (not here in email)?**** >> >> ** ** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Burak >> BorYazilim >> *Sent:* Saturday, May 11, 2013 7:28 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Freeswitch User Adding**** >> >> ** ** >> >> First off all thanks for your kind helps.**** >> >> ** ** >> >> Let me tell you the situation more clearly.**** >> >> ** ** >> >> Firstly I used Android phones as sip clients with the open source sip >> client program, CSipSimple. My server computer has Ubuntu 12.04 LTS with >> one static ip address.**** >> >> freeswitch version output: FreeSWITCH Version >> 1.3.13b+git~20130205T003128Z~70a9560306 (git 70a9560 2013-02-05 00:31:28Z) >> **** >> >> We were using the system with 4 digits number sip account names without >> any problem. Also the account name will be used as dial number. An example >> of the user xml file is below. I changed the regex in dialpan/default.xml. >> The changings were perfectly succesfull with, again, 4 digits number. But >> when trying to include any other number of digits (3, 5 and 6 were tested), >> it doesn't work(When I change the regex to accept only 5 digits numbers, 4 >> digits ones didnt work as expected). I really could not understand why it >> is happening. Why there was no problem with 4 digits number and why the >> exact same system does not work with this basic change.**** >> >> ** ** >> >> To explain the errors more, I want to talk about my tests. Firstly I used >> 4 digits number user. This test repeated with tls and srtp. These two test >> were succesfull. **** >> >> And the secand test is same with first test but with 5 digits numbers >> without tls and srtp. Registration was succesfull but cant call. (or the >> call could not be forwarded) Of course I changed the regex and execute >> reloadxml in this test. GSM (8kHz) and G722(16kHz) codecs were used. As you >> see below, CODEC NEGOTIATION ERROR occured. After getting this error I >> changed the codes as SILK(16kHz). With this change I made the third test. >> Third test result is also below. **** >> >> ** ** >> >> show registretions output: **** >> >> >> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata >> **** >> >> >> 60022,my.server.ip.address,oOnxiVJFAgQWLABrD01UsYTVOY3TVSlx,sofia/internal/sip:60022 at 141.196.174.60:52245 >> ;ob,1368271435,141.196.174.60,52245,udp,server,**** >> >> >> 60021,my.server.ip.address,5r4MTDiZPhs5qzdin9A3hEUh1zZsdqqk,sofia/internal/sip:60021 at 141.196.174.60:57938 >> ;ob,1368271446,141.196.174.60,57938,udp,server,**** >> >> ** ** >> >> ** ** >> >> @Sean, sorry but I cant post my whole log file because my server has got >> a network attack so I am quite busy with this attacker. And also I dont >> think CSipSimple has a problem with 5 digits because this system (with 5 >> digits) was succesfull with Kamailio and CSipSimple.**** >> >> ** ** >> >> If I could not be clear please let me know. Thanks...**** >> >> ** ** >> >> user xml example: This was exact same with the default users in >> freeswitch. Only the number changed.**** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> * >> *** >> >> **** >> >> **** >> >> >> **** >> >> **** >> >> > value="$${outbound_caller_name}"/>**** >> >> > value="$${outbound_caller_id}"/>**** >> >> **** >> >> **** >> >> **** >> >> **** >> >> ** ** >> >> ** ** >> >> ** ** >> >> The second test error:**** >> >> ** ** >> >> 2013-05-11 12:13:56.800443 [DEBUG] sofia_reg.c:1511 Send challenge for >> [60021 at my.server.ip.address]**** >> >> 2013-05-11 12:13:56.900444 [DEBUG] sofia_reg.c:1511 Send challenge for >> [60021 at my.server.ip.address] >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'record_stereo' = 'true' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_gateway' = 'example.com' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_areacode' = '918' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'transfer_fallback_extension' = 'operator' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'toll_allow' = 'domestic,international,local' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'accountcode' = '60021' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'user_context' = 'default' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_name' = 'Extension 60021' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_number' = '60021' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_name' = 'FreeSWITCH' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_number' = '0000000000' >> **** >> >> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'callgroup' = 'techsupport' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'record_stereo' = 'true' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_gateway' = 'example.com' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_areacode' = '918' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'transfer_fallback_extension' = 'operator' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'toll_allow' = 'domestic,international,local' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'accountcode' = '60021' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'user_context' = 'default' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_name' = 'Extension 60021' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_number' = '60021' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_name' = 'FreeSWITCH' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_number' = '0000000000' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'callgroup' = 'techsupport' >> **** >> >> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:1683 Register: >> >> **** >> >> From: [60021 at my.server.ip.address] >> >> **** >> >> Contact: ["user" ] >> >> **** >> >> Expires: [900] >> >> **** >> >> 2013-05-11 12:14:06.820443 [DEBUG] sofia_reg.c:1511 Send challenge for >> [60022 at my.server.ip.address] >> **** >> >> 2013-05-11 12:14:06.900444 [DEBUG] sofia_reg.c:1511 Send challenge for >> [60022 at my.server.ip.address] >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'record_stereo' = 'true' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_gateway' = 'example.com' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_areacode' = '918' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'transfer_fallback_extension' = 'operator' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'toll_allow' = 'domestic,international,local' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'accountcode' = '60022' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'user_context' = 'default' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_name' = 'Extension 60022' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_number' = '60022' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_name' = 'FreeSWITCH' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_number' = '0000000000' >> **** >> >> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'callgroup' = 'techsupport' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'record_stereo' = 'true' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_gateway' = 'example.com' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_areacode' = '918' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'transfer_fallback_extension' = 'operator' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'toll_allow' = 'domestic,international,local' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'accountcode' = '60022' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'user_context' = 'default' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_name' = 'Extension 60022' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_number' = '60022' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_name' = 'FreeSWITCH' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_number' = '0000000000' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'callgroup' = 'techsupport' >> **** >> >> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:1683 Register: >> >> **** >> >> From: [60022 at my.server.ip.address] >> >> **** >> >> Contact: ["user" ] >> >> **** >> >> Expires: [900] >> >> **** >> >> 2013-05-11 12:14:14.640437 [NOTICE] switch_channel.c:968 New Channel >> sofia/internal/60021 at my.server.ip.address[26730cd2-ba1b-11e2-acc5-bda7cbfd9554] >> **** >> >> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK]**** >> >> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:415 ( >> sofia/internal/60021 at my.server.ip.address) Running State Change CS_NEW >> **** >> >> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:433 ( >> sofia/internal/60021 at my.server.ip.address) State NEW >> **** >> >> 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:7733 IP 141.196.174.60 >> Rejected by acl "domains". Falling back to Digest auth. >> **** >> >> 2013-05-11 12:14:14.660438 [DEBUG] sofia_reg.c:1511 Send challenge for >> [60022 at my.server.ip.address] >> **** >> >> 2013-05-11 12:14:14.660438 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:1719 detaching session >> 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 >> **** >> >> 2013-05-11 12:14:14.780439 [DEBUG] sofia.c:1811 Re-attaching to session >> 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 >> **** >> >> 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:7733 IP 141.196.174.60 >> Rejected by acl "domains". Falling back to Digest auth. >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'record_stereo' = 'true' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_gateway' = 'example.com' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'default_areacode' = '918' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'transfer_fallback_extension' = 'operator' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'toll_allow' = 'domestic,international,local' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'accountcode' = '60021' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'user_context' = 'default' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_name' = 'Extension 60021' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'effective_caller_id_number' = '60021' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_name' = 'FreeSWITCH' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'outbound_caller_id_number' = '0000000000' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >> 'callgroup' = 'techsupport' >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel >> sofia/internal/60021 at my.server.ip.address entering state [received][100] >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5589 Remote SDP: >> >> **** >> >> v=0 >> >> **** >> >> o=- 3577252345 3577252345 IN IP4 141.196.174.60 >> >> **** >> >> s=pjmedia >> >> **** >> >> c=IN IP4 141.196.174.60 >> >> **** >> >> t=0 0 >> >> **** >> >> m=audio 4010 RTP/AVP 8 0 3 101 >> >> **** >> >> c=IN IP4 141.196.174.60 >> >> **** >> >> a=rtpmap:8 PCMA/8000 >> >> **** >> >> a=rtpmap:0 PCMU/8000 >> >> **** >> >> a=rtpmap:3 GSM/8000 >> >> **** >> >> a=rtpmap:101 telephone-event/8000 >> >> **** >> >> a=fmtp:101 0-15 >> >> **** >> >> a=rtcp:4011 IN IP4 192.168.43.10 >> >> **** >> >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >> **** >> >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >> **** >> >> ** ** >> >> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5802 ( >> sofia/internal/60021 at my.server.ip.address) State Change CS_NEW -> >> CS_INIT **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( >> sofia/internal/60021 at my.server.ip.address) Running State Change CS_INIT >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( >> sofia/internal/60021 at my.server.ip.address) State INIT >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 >> sofia/internal/60021 at my.server.ip.address SOFIA INIT >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 ( >> sofia/internal/60021 at my.server.ip.address) State Change CS_INIT -> >> CS_ROUTING **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send signal >> sofia/internal/60021 at my.server.ip.address [BREAK] >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( >> sofia/internal/60021 at my.server.ip.address) State INIT going to sleep >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( >> sofia/internal/60021 at my.server.ip.address) Running State Change >> CS_ROUTING **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 ( >> sofia/internal/60021 at my.server.ip.address) Callstate Change DOWN -> >> RINGING **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 ( >> sofia/internal/60021 at my.server.ip.address) State ROUTING >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 >> sofia/internal/60021 at my.server.ip.address SOFIA ROUTING >> **** >> >> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:117 >> sofia/internal/60021 at my.server.ip.address Standard ROUTING >> **** >> >> 2013-05-11 12:14:14.800439 [INFO] mod_dialplan_xml.c:557 Processing 60021 >> <60021>->60022 in context default**** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->unloop] continue=false**** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->tod_example] continue=true >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch >> (FAIL) [tod_example] break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->holiday_example] continue=true >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch >> (FAIL) [holiday_example] break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->global-intercept] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [global-intercept] destination_number(60022) =~ /^886$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->group-intercept] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [group-intercept] destination_number(60022) =~ /^\*8$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->intercept-ext] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [intercept-ext] destination_number(60022) =~ /^\*\*(\d+)$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->redial] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [redial] destination_number(60022) =~ /^(redial|870)$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->global] continue=true >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [global] ${call_debug}(false) =~ /^true$/ break=never >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [global] ${sip_has_crypto}() =~ >> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >> [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED >> NEGOTIATION)/ break=on-false **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >> [global] ${switch_r_sdp}(v=0 >> **** >> >> o=- 3577252345 3577252345 IN IP4 141.196.174.60 >> >> **** >> >> s=pjmedia >> >> **** >> >> c=IN IP4 141.196.174.60 >> >> **** >> >> t=0 0 >> >> **** >> >> m=audio 4010 RTP/AVP 8 0 3 101 >> >> **** >> >> c=IN IP4 141.196.174.60 >> >> **** >> >> a=rtpmap:8 PCMA/8000 >> >> **** >> >> a=rtpmap:0 PCMU/8000 >> >> **** >> >> a=rtpmap:3 GSM/8000 >> >> **** >> >> a=rtpmap:101 telephone-event/8000 >> >> **** >> >> a=fmtp:101 0-15 >> >> **** >> >> a=rtcp:4011 IN IP4 192.168.43.10 >> >> **** >> >> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >> **** >> >> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >> **** >> >> ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never >> >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(sip_secure_media=true) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> export(sip_secure_media=true) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Absolute Condition >> [global] >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-last_dial/global/${uuid}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->snom-demo-2] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [snom-demo-2] destination_number(60022) =~ /^9001$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->snom-demo-1] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [snom-demo-1] destination_number(60022) =~ /^9000$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->eavesdrop] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [eavesdrop] destination_number(60022) =~ /^88(\d{4})$|^\*0(.*)$/ >> break=on-false **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->eavesdrop] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [eavesdrop] destination_number(60022) =~ /^779$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->call_return] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [call_return] destination_number(60022) =~ /^\*69$|^869$|^lcr$/ >> break=on-false **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->del-group] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [del-group] destination_number(60022) =~ /^80(\d{2})$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->add-group] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [add-group] destination_number(60022) =~ /^81(\d{2})$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->call-group-simo] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [call-group-simo] destination_number(60022) =~ /^82(\d{2})$/ break=on-false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->call-group-order] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [call-group-order] destination_number(60022) =~ /^83(\d{2})$/ >> break=on-false **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->extension-intercom] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >> [extension-intercom] destination_number(60022) =~ /^8(10[01][0-9])$/ >> break=on-false **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >> [default->Local_Extension] continue=false >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >> [Local_Extension] destination_number(60022) =~ >> /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false >> >> >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> export(dialed_extension=60022) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> bind_meta_app(1 b s execute_extension::dx XML features) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> bind_meta_app(2 b s >> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) >> >> >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> bind_meta_app(3 b s execute_extension::cf XML features) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> bind_meta_app(4 b s execute_extension::att_xfer XML features) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(ringback=${us-ring}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(transfer_ringback=local_stream://moh) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(call_timeout=30) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(hangup_after_bridge=true) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(continue_on_fail=true) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >> var callgroup)}) **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) >> **** >> >> Dialplan: sofia/internal/60021 at my.server.ip.address Action >> hash(insert/${domain_name}-last_dial_ext/global/${uuid}) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> ... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/11debadf/attachment-0001.html From ashish at nms.co.in Mon May 13 11:20:26 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 13 May 2013 12:50:26 +0530 Subject: [Freeswitch-users] hangup extension Message-ID: Hi, Is there anything like hangup extension which gets executed after the call is hung up for which the UUID is same and based on that I can do things for the same call like we have in Asterisk? Thanks. Regards, --Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/d0ea9b19/attachment.html From steveayre at gmail.com Mon May 13 11:49:08 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 08:49:08 +0100 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: The call duration is 0 seconds so no they didn't get in. Any accepted call attempt will generate a CDR even if it was not answered. They're coming in on the 'public' context which'll mean an unauthenticated call on the external profile. That would be the context allowing you to receive calls from strangers. Providing that context doesn't allow them to dial out via providers that charge you this isn't an issue. -Steve On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > I have lot of entries like 1000,1001,1002 and so on, Is someone trying to > get inside? I believe they didn't in yet but how come we have cdr log like > this? > > "1004","1004","012972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","013972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","014972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","010972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0061972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0041972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","000972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","006972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","002972598371070","public","2013-05-12 > 07:30:10","","2013-05-12 > 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","810972598371070","public","2013-05-12 > 07:29:49","","2013-05-12 > 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/9a433f04/attachment.html From b2m at a-cti.com Mon May 13 12:02:10 2013 From: b2m at a-cti.com (Bala Murugan Mahendran) Date: Mon, 13 May 2013 13:32:10 +0530 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: Thanks for your help. Do I still need to do any precautionary change? Thanks, Bala On Mon, May 13, 2013 at 1:19 PM, Steven Ayre wrote: > The call duration is 0 seconds so no they didn't get in. Any accepted call > attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an unauthenticated > call on the external profile. That would be the context allowing you to > receive calls from strangers. Providing that context doesn't allow them to > dial out via providers that charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > >> I have lot of entries like 1000,1001,1002 and so on, Is someone trying to >> get inside? I believe they didn't in yet but how come we have cdr log like >> this? >> >> "1004","1004","012972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","013972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","014972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","010972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0061972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0041972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","000972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","006972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","002972598371070","public","2013-05-12 >> 07:30:10","","2013-05-12 >> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","810972598371070","public","2013-05-12 >> 07:29:49","","2013-05-12 >> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" >> >> >> Thanks, >> Bala >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/cd227c3e/attachment.html From jeff at askcornerstone.net Mon May 13 12:03:56 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Mon, 13 May 2013 08:03:56 +0000 Subject: [Freeswitch-users] Recommendations for Trunks from Hawaii? Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> Not Freeswitch specific, but... We have a couple Vitelity DIDs we use for testing, but the latency on them is fairly bad. We figure this is just because of physical distance... we're in Hawaii and Vitelity is I believe in Denver (at least that's the ip we connect to). Does anyone have recommendations for any good trunks that might be located somewhere closer (California)? There are plenty of companies here and there, but what are people using that might meet our needs? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/9d9df326/attachment-0001.html From steveayre at gmail.com Mon May 13 15:07:32 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 12:07:32 +0100 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: If you're concerned someone's trying to break in eg by guessing passwords you could look at http://wiki.freeswitch.org/wiki/Mod_fail2ban. A number of failed login attempts will block them from connecting again. Also check over your configuration to check what they would be able to do from the public context, and verify you're not using any of the default passwords. -Steve On 13 May 2013 09:02, Bala Murugan Mahendran wrote: > Thanks for your help. Do I still need to do any precautionary change? > > Thanks, > Bala > > > On Mon, May 13, 2013 at 1:19 PM, Steven Ayre wrote: > >> The call duration is 0 seconds so no they didn't get in. Any accepted >> call attempt will generate a CDR even if it was not answered. >> >> They're coming in on the 'public' context which'll mean an >> unauthenticated call on the external profile. That would be the context >> allowing you to receive calls from strangers. Providing that context >> doesn't allow them to dial out via providers that charge you this isn't an >> issue. >> >> -Steve >> >> >> >> >> >> >> On 13 May 2013 07:17, Bala Murugan Mahendran wrote: >> >>> I have lot of entries like 1000,1001,1002 and so on, Is someone trying >>> to get inside? I believe they didn't in yet but how come we have cdr log >>> like this? >>> >>> "1004","1004","012972598371070","public","2013-05-12 >>> 07:30:07","","2013-05-12 >>> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","013972598371070","public","2013-05-12 >>> 07:30:07","","2013-05-12 >>> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","014972598371070","public","2013-05-12 >>> 07:30:07","","2013-05-12 >>> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","010972598371070","public","2013-05-12 >>> 07:30:08","","2013-05-12 >>> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","0061972598371070","public","2013-05-12 >>> 07:30:08","","2013-05-12 >>> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","0041972598371070","public","2013-05-12 >>> 07:30:09","","2013-05-12 >>> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","000972598371070","public","2013-05-12 >>> 07:30:09","","2013-05-12 >>> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","006972598371070","public","2013-05-12 >>> 07:30:09","","2013-05-12 >>> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","002972598371070","public","2013-05-12 >>> 07:30:10","","2013-05-12 >>> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" >>> "1004","1004","810972598371070","public","2013-05-12 >>> 07:29:49","","2013-05-12 >>> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" >>> >>> >>> Thanks, >>> Bala >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/9ce61687/attachment.html From bdfoster at davri.com Mon May 13 16:00:21 2013 From: bdfoster at davri.com (Brian Foster) Date: Mon, 13 May 2013 08:00:21 -0400 Subject: [Freeswitch-users] Recommendations for Trunks from Hawaii? In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> References: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> Message-ID: Flowroute is in Las Vegas and I think Los Angeles. Might wantvto check them out: http://flowroute.com -BDF On May 13, 2013 4:11 AM, "Jeff Bernhardt" wrote: > Not Freeswitch specific, but?**** > > We have a couple Vitelity DIDs we use for testing, but the latency on them > is fairly bad. We figure this is just because of physical distance? we?re > in Hawaii and Vitelity is I believe in Denver (at least that?s the ip we > connect to). Does anyone have recommendations for any good trunks that > might be located somewhere closer (California)? There are plenty of > companies here and there, but what are people using that might meet our > needs?**** > > Thanks.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/d2e12655/attachment.html From b2m at a-cti.com Mon May 13 16:03:05 2013 From: b2m at a-cti.com (Bala Murugan Mahendran) Date: Mon, 13 May 2013 17:33:05 +0530 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: Yes, I already have this. Looks everything is okay. Thanks for your help!! Thanks, Bala On Mon, May 13, 2013 at 4:37 PM, Steven Ayre wrote: > If you're concerned someone's trying to break in eg by guessing passwords > you could look at http://wiki.freeswitch.org/wiki/Mod_fail2ban. A number > of failed login attempts will block them from connecting again. > > Also check over your configuration to check what they would be able to do > from the public context, and verify you're not using any of the default > passwords. > > -Steve > > > > > On 13 May 2013 09:02, Bala Murugan Mahendran wrote: > >> Thanks for your help. Do I still need to do any precautionary change? >> >> Thanks, >> Bala >> >> >> On Mon, May 13, 2013 at 1:19 PM, Steven Ayre wrote: >> >>> The call duration is 0 seconds so no they didn't get in. Any accepted >>> call attempt will generate a CDR even if it was not answered. >>> >>> They're coming in on the 'public' context which'll mean an >>> unauthenticated call on the external profile. That would be the context >>> allowing you to receive calls from strangers. Providing that context >>> doesn't allow them to dial out via providers that charge you this isn't an >>> issue. >>> >>> -Steve >>> >>> >>> >>> >>> >>> >>> On 13 May 2013 07:17, Bala Murugan Mahendran wrote: >>> >>>> I have lot of entries like 1000,1001,1002 and so on, Is someone trying >>>> to get inside? I believe they didn't in yet but how come we have cdr log >>>> like this? >>>> >>>> "1004","1004","012972598371070","public","2013-05-12 >>>> 07:30:07","","2013-05-12 >>>> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","013972598371070","public","2013-05-12 >>>> 07:30:07","","2013-05-12 >>>> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","014972598371070","public","2013-05-12 >>>> 07:30:07","","2013-05-12 >>>> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","010972598371070","public","2013-05-12 >>>> 07:30:08","","2013-05-12 >>>> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","0061972598371070","public","2013-05-12 >>>> 07:30:08","","2013-05-12 >>>> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","0041972598371070","public","2013-05-12 >>>> 07:30:09","","2013-05-12 >>>> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","000972598371070","public","2013-05-12 >>>> 07:30:09","","2013-05-12 >>>> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","006972598371070","public","2013-05-12 >>>> 07:30:09","","2013-05-12 >>>> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","002972598371070","public","2013-05-12 >>>> 07:30:10","","2013-05-12 >>>> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" >>>> "1004","1004","810972598371070","public","2013-05-12 >>>> 07:29:49","","2013-05-12 >>>> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" >>>> >>>> >>>> Thanks, >>>> Bala >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/5b3740fc/attachment-0001.html From john at millican.us Mon May 13 16:55:20 2013 From: john at millican.us (john at millican.us) Date: Mon, 13 May 2013 08:55:20 -0400 Subject: [Freeswitch-users] Recommendations for Trunks from Hawaii? In-Reply-To: References: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> Message-ID: <5190E2B8.5060306@millican.us> On 5/13/2013 8:00 AM, Brian Foster wrote: > > Flowroute is in Las Vegas and I think Los Angeles. Might wantvto check > them out: http://flowroute.com > > -BDF > > On May 13, 2013 4:11 AM, "Jeff Bernhardt" > wrote: > > Not Freeswitch specific, but... > > We have a couple Vitelity DIDs we use for testing, but the latency > on them is fairly bad. We figure this is just because of physical > distance... we're in Hawaii and Vitelity is I believe in Denver > (at least that's the ip we connect to). Does anyone have > recommendations for any good trunks that might be located > somewhere closer (California)? There are plenty of companies here > and there, but what are people using that might meet our needs? > > Thanks. > > Speaking based only on "assumptions" here but I doubt if there will be much difference between Hawaii and a SIP provider in LA or one in Denver. I have had servers of my own in New Hampshire, Virginia, and LA and the max ping times over public Internet were typically around 95ms. I would bet that the majority of the latency you are seeing is due to the link between Hawaii and the mainland. Could it be a satellite link that your Internet provider is using? Have you asked them about the connectivity they use to get to mainland destinations? JohnM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/0bd46185/attachment.html From krice at freeswitch.org Mon May 13 18:35:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 09:35:15 -0500 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: Message-ID: Ok look at the 3rd column... This is someone trying to figure out how to get calls thru your public context with various prefix codexs... Notices how all the calls end with 972598371070 <-- that?s the intl destination umber probably a phone in gaza (definitely in Isreali controlled areas, but I think I looked up this number or similar number in the past) To protect yourself, only allow calls to local extentsions from the public context... On 5/13/13 2:49 AM, "Steven Ayre" wrote: > The call duration is 0 seconds so no they didn't get in. Any accepted call > attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an unauthenticated > call on the external profile. That would be the context allowing you to > receive calls from strangers. Providing that context doesn't allow them to > dial out via providers that charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: >> I have lot of?entries?like 1000,1001,1002 and so on, Is someone trying to get >> inside? I?believe?they didn't in yet but how come we have cdr log like this? >> >> "1004","1004","012972598371070","public","2013-05-12 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","013972598371070","public","2013-05-12 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","014972598371070","public","2013-05-12 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","010972598371070","public","2013-05-12 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","0061972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","0041972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","000972598371070","public","2013-05-12 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","006972598371070","public","2013-05-12 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","002972598371070","public","2013-05-12 07:30:10","","2013-05-12 >> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","" >> ,"","G729","G729" >> "1004","1004","810972598371070","public","2013-05-12 07:29:49","","2013-05-12 >> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","" >> ,"","G729","G729" >> >> >> Thanks, >> Bala >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/c6312a5d/attachment.html From emamirazavi at gmail.com Mon May 13 18:37:04 2013 From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi) Date: Mon, 13 May 2013 19:07:04 +0430 Subject: [Freeswitch-users] An strange bug on starting up FreeSwitch, any solution? Message-ID: Hello, I downloaded FS version 1.5.0 () and made it again. my previous version was 1.3.13b+git~20130114T032717Z~d78f4ffb19 Now my problem has been solved. At mentioned server i had downloaded ffmpeg from dag resource(CentOS package source list) and after that FS went down! but with new FS the problem currently has solved! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/f634f78b/attachment.html From steveayre at gmail.com Mon May 13 19:43:01 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 16:43:01 +0100 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: +1. This is what i meant by checking the public context. :) On 13 May 2013 15:35, Ken Rice wrote: > Ok look at the 3rd column... This is someone trying to figure out how to > get calls thru your public context with various prefix codexs... Notices > how all the calls end with 972598371070 <-- that?s the intl destination > umber probably a phone in gaza (definitely in Isreali controlled areas, but > I think I looked up this number or similar number in the past) > > To protect yourself, only allow calls to local extentsions from the public > context... > > > On 5/13/13 2:49 AM, "Steven Ayre" wrote: > > The call duration is 0 seconds so no they didn't get in. Any accepted call > attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an unauthenticated > call on the external profile. That would be the context allowing you to > receive calls from strangers. Providing that context doesn't allow them to > dial out via providers that charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > > I have lot of entries like 1000,1001,1002 and so on, Is someone trying to > get inside? I believe they didn't in yet but how come we have cdr log like > this? > > "1004","1004","012972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","013972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","014972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","010972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0061972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0041972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","000972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","006972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","002972598371070","public","2013-05-12 > 07:30:10","","2013-05-12 > 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","810972598371070","public","2013-05-12 > 07:29:49","","2013-05-12 > 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/b50973ab/attachment-0001.html From jh.zhou at outlook.com Mon May 13 09:43:05 2013 From: jh.zhou at outlook.com (ZhouJianhua) Date: Mon, 13 May 2013 05:43:05 +0000 Subject: [Freeswitch-users] freeswitch's db error: took too long to complete Message-ID: Hi all, I installed freeswitch and postgresql on amazon ec2, after 1 day or 2 days, the following error occurred: switch_pgsql.c:359 Query (delete from interfaces where type='api' and name='start_local_stream' and hostname='fs01') took too long to complete or database not responding. I have to shutdown freeswitch and restart it. There is no problem on our physic machine with same configuration, only happened on amazon ec2.Anyone encountered similar problems ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/18944168/attachment.html From jcho at radixsystems.com Mon May 13 12:39:04 2013 From: jcho at radixsystems.com (jimmy cho) Date: Mon, 13 May 2013 16:39:04 +0800 Subject: [Freeswitch-users] zrtp not working Message-ID: Hi, I am using the default configurations in the freeswitch server. I have enabled zrtp on a csimplesip client and a twinkle client. If I configure the freeswitch server with The csimplesip client can connect with zrtp with the twinkle client on a linux desktop. ( I get the lock icon ) ( same network ) As my clients can be on different subnets, bypass is not suitable. How with the bypass media parameter commented out I get this is in the freeswitch log --> ZRTP not negotiated on both sides; disabling ZRTP passthru mode. Also the twinkle has the log entry WARNING NORMAL TwinkleZrtpUI::showMessage Line 1: Internal ZRTP packet checksum mismatch - packet dropped I do have enabled. Do I need a zrtp_entrollment=true set somewhere. If this is reqiured, where should it be configured ? Any suggestions.. Many Thanks Jimmy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/f6758aa4/attachment.html From ostolyar at netflix.com Mon May 13 19:40:00 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Mon, 13 May 2013 08:40:00 -0700 Subject: [Freeswitch-users] Dynamic external profile proxies Message-ID: I am very new to FreeSWITCH and would like to use it as a SIP load balancer to other (media) FreeSWITCH instances. I was thinking of using the external SIP profile and bridging incoming calls to the media instances of FreeSWITCH using some rotation mechanism (round robin, least busy, etc) My question is this: Would I need to have a separate SIP profile for each media FreeSWITCH server or can I define a profile in such a way that I can dynamically specify the proxy for each call? Thank you *Oleg* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/37fcc5fa/attachment.html From tru083 at yahoo.com Mon May 13 19:54:19 2013 From: tru083 at yahoo.com (D D) Date: Mon, 13 May 2013 08:54:19 -0700 (PDT) Subject: [Freeswitch-users] eavesdrop not sending video Message-ID: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> Hi, I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. I have followed the directions described at?http://wiki.freeswitch.org/wiki/Video-recording (including a rebuild after doing the git checkout of?video-media-bug),?and I can successfully? use?record_fsv to record the video to a file. Any ideas why the eavesdrop does not send the video? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/2fa15b63/attachment.html From john at telecube.com.au Mon May 13 09:44:25 2013 From: john at telecube.com.au (Telecube - John) Date: Mon, 13 May 2013 15:44:25 +1000 Subject: [Freeswitch-users] FW: failed registrations In-Reply-To: References: Message-ID: <51907DB9.2030408@telecube.com.au> On 13/05/13 10:00 AM, Tom Berchenbriter wrote: > > > I have this freeswitch box im trying to get registered to my zoiper > sip client, with no success... Its reaching the box and the service, > but I cant get it to register. Please help! > > It would appear to be simply password error. > 2013-05-12 16:39:05.798483 [WARNING] sofia_reg.c:1390 SIP auth failure (REGISTER) on sofia profile 'internal' for [tom at freeswitch.vunity.com] from ip 76.114.1.252 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/d787b01b/attachment-0001.html From john at telecube.com.au Mon May 13 13:04:10 2013 From: john at telecube.com.au (Telecube - John) Date: Mon, 13 May 2013 19:04:10 +1000 Subject: [Freeswitch-users] hangup extension In-Reply-To: References: Message-ID: <5190AC8A.1080903@telecube.com.au> If you are using Lua script there is ' session:setHangupHook' detailed here: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook I assume there's something similar in the xml configs Regards, John On 13/05/13 5:20 PM, Ashish gautam wrote: > Hi, > > Is there anything like hangup extension which gets executed after the > call is hung up for which the UUID is same and based on that I can do > things for the same call like we have in Asterisk? > > Thanks. > > Regards, > > --Ashish > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/b2f1235e/attachment-0001.html From john at telecube.com.au Mon May 13 18:52:06 2013 From: john at telecube.com.au (Telecube - John) Date: Tue, 14 May 2013 00:52:06 +1000 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: <5190FE16.6090502@telecube.com.au> You can install a log watching program that looks out for specific log entries and can take action. Eg; writing a firewall rule to ban the ip address of the system trying to make unauthorised calls. 'swatch' is a good tool for this in linux systems. http://sourceforge.net/projects/swatch/ John On 14/05/13 12:35 AM, Ken Rice wrote: > Re: [Freeswitch-users] Strange CDR entry Ok look at the 3rd column... > This is someone trying to figure out how to get calls thru your public > context with various prefix codexs... Notices how all the calls end > with 972598371070 <-- that's the intl destination umber probably a > phone in gaza (definitely in Isreali controlled areas, but I think I > looked up this number or similar number in the past) > > To protect yourself, only allow calls to local extentsions from the > public context... > > > On 5/13/13 2:49 AM, "Steven Ayre" wrote: > > The call duration is 0 seconds so no they didn't get in. Any > accepted call attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an > unauthenticated call on the external profile. That would be the > context allowing you to receive calls from strangers. Providing > that context doesn't allow them to dial out via providers that > charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > > I have lot of entries like 1000,1001,1002 and so on, Is > someone trying to get inside? I believe they didn't in yet but > how come we have cdr log like this? > > "1004","1004","012972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","013972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","014972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","010972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0061972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0041972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","000972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","006972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","002972598371070","public","2013-05-12 > 07:30:10","","2013-05-12 > 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","810972598371070","public","2013-05-12 > 07:29:49","","2013-05-12 > 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/693f4214/attachment-0001.html From krice at freeswitch.org Mon May 13 20:17:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 11:17:41 -0500 Subject: [Freeswitch-users] freeswitch's db error: took too long to complete In-Reply-To: Message-ID: Sounds like disk io issues... Here in lies the problem with ?cloud? virtualization... You have no control over what else is running on the shared hardware... You are also probably running a small instance... Check out this mailing list and other freeswitch related discussions on the web about virtualiztion on EC2 On 5/13/13 12:43 AM, "ZhouJianhua" wrote: > Hi all, > > I installed freeswitch and postgresql on amazon ec2, after 1 day or 2 days, > the following error occurred: > > switch_pgsql.c:359 Query (delete from interfaces where type='api' and > name='start_local_stream' and hostname='fs01') took too long to complete or > database not responding. > > I have to shutdown freeswitch and restart it. There is no problem on our > physic machine with same configuration, only happened on amazon ec2. > Anyone encountered similar problems ? > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/a2a9d561/attachment.html From krice at freeswitch.org Mon May 13 20:19:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 11:19:08 -0500 Subject: [Freeswitch-users] eavesdrop not sending video In-Reply-To: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> Message-ID: Eavesdrop probably does not support sending the video stream as well... If I recall correctly, it only taps the audio stream... Possible solutions, open a jira and offer a bounty for someone to look at adding support for video On 5/13/13 10:54 AM, "D D" wrote: > Hi, > > I tried to use eavesdrop to capture the video of a parked call, but I only > receive the audio. > > I have followed the directions described at > http://wiki.freeswitch.org/wiki/Video-recording > (including a rebuild after doing the git checkout of video-media-bug), and I > can successfully > use record_fsv to record the video to a file. > > Any ideas why the eavesdrop does not send the video? > > Thanks, > David > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/3d8fe3b8/attachment.html From steveayre at gmail.com Mon May 13 20:28:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 17:28:48 +0100 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: <5190FE16.6090502@telecube.com.au> References: <5190FE16.6090502@telecube.com.au> Message-ID: This is the purpose of the fail2ban system that was already mentioned. -Steve On 13 May 2013 15:52, Telecube - John wrote: > You can install a log watching program that looks out for specific log > entries and can take action. Eg; writing a firewall rule to ban the ip > address of the system trying to make unauthorised calls. > > 'swatch' is a good tool for this in linux systems. > > http://sourceforge.net/projects/swatch/ > > John > > On 14/05/13 12:35 AM, Ken Rice wrote: > > Ok look at the 3rd column... This is someone trying to figure out how to > get calls thru your public context with various prefix codexs... Notices > how all the calls end with 972598371070 <-- that?s the intl destination > umber probably a phone in gaza (definitely in Isreali controlled areas, but > I think I looked up this number or similar number in the past) > > To protect yourself, only allow calls to local extentsions from the public > context... > > > On 5/13/13 2:49 AM, "Steven Ayre" wrote: > > The call duration is 0 seconds so no they didn't get in. Any accepted > call attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an unauthenticated > call on the external profile. That would be the context allowing you to > receive calls from strangers. Providing that context doesn't allow them to > dial out via providers that charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > > I have lot of entries like 1000,1001,1002 and so on, Is someone trying to > get inside? I believe they didn't in yet but how come we have cdr log like > this? > > "1004","1004","012972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","013972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","014972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","010972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0061972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0041972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","000972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","006972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","002972598371070","public","2013-05-12 > 07:30:10","","2013-05-12 > 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","810972598371070","public","2013-05-12 > 07:29:49","","2013-05-12 > 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130513/5acaba76/attachment-0001.html From alex at digitalmail.com Mon May 13 21:21:54 2013 From: alex at digitalmail.com (Alex Lake) Date: Mon, 13 May 2013 18:21:54 +0100 Subject: [Freeswitch-users] Changes between versions Message-ID: <51912132.4080407@digitalmail.com> Is there a clever way in which one can use git/github to tell me (in human readable terms!) all the changes between version A and version B of Freeswitch? From schoch+freeswitch.org at xwin32.com Mon May 13 21:52:11 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 13 May 2013 10:52:11 -0700 Subject: [Freeswitch-users] Caller ID - a moving target In-Reply-To: References: <11ff01ce4db9$20397cd0$60ac7670$@bizfocused.com> <138701ce4de7$4fad61b0$ef082510$@bizfocused.com> Message-ID: My issue with internal calls is that when the called extension is answered, the CID display of the station that originated the call (the caller), changes the CID name from the extension number to "Outbound Call". It would be nice if it could show the actual name. -- Steve > When someone dials my extension from an internal extension, my phone >> displays the CID Number. When I answer, the display switches to CID Name. >> >> ** >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/38a2c645/attachment.html From krice at freeswitch.org Mon May 13 21:58:26 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 12:58:26 -0500 Subject: [Freeswitch-users] Changes between versions In-Reply-To: <51912132.4080407@digitalmail.com> Message-ID: Git log tag..tag I think On 5/13/13 12:21 PM, "Alex Lake" wrote: > Is there a clever way in which one can use git/github to tell me (in > human readable terms!) all the changes between version A and version B > of Freeswitch? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From asaad2 at gmail.com Mon May 13 22:05:54 2013 From: asaad2 at gmail.com (BookBag) Date: Mon, 13 May 2013 14:05:54 -0400 Subject: [Freeswitch-users] Changes between versions In-Reply-To: <51912132.4080407@digitalmail.com> References: <51912132.4080407@digitalmail.com> Message-ID: -----BEGIN PGP PUBLIC KEY BLOCK----- Version: OpenPGP.js v.1.20130228 Comment: http://openpgpjs.org xsFNBFGERFIBEACeX3R6n+D3jK8KluJj0fTZOazMAh5JRdBDjeqI139T+uln SzVeVXcgb7Gs3Ag5Ugo9S+nqPWGmuUOC1w39LMpDixYu7G5Cy1CG4wtw3ALR kixYvqNPEHA4vwBq1jSLcRWeXNgrd97aWGltYXYjI7ZorPrWefLGJp7/24Sf iUyCtD37EXYGX4a34lmbjH20qhZCAONlC9wyXdvJLkm4Nu4yQC4DujC7wWwQ Wj8AFm6IZiqVuJTs8B9bdLrK3a1KgpZggSq1zWHppljplQseWsKz/4aXbOtY v2BpCCyFBscFJhaoFpATRpY1RT3PT6Jf4JJiBK0UWp9akKycp5D8/+cetO/4 bekXQP7dXtooV9+e1iwCccg+/cZaGUKChry+C1wMbe/8pW3KgaYq2SCIUFjh +Q07CVEkwvkNHgrUy48FI9eQQ7cKnaIF+3z8Rh2OkxczB95ym1saN62WqAkK bF1wpDGxFSkTiT60k6ioFqE2Lo+F+2JFA3Acpyp2gLsHnpT4/pQ5viSp+3ph 5i7ZosoWli+JHNFGJL6P3RU+QoUJQ/carcBhGSx4xhgo9gNjQU8ImiJbbWfp aMX9YmuKvIp4ZsslkDhWWUpkMgBTb97SfKDvwZ8zaQQAC2KKFsH9fVmPFAvu J1CJ858NCU7Ys04R91kfK5EdBkA+z5vYmdlwVQARAQABzSRNb2hhbW1lZCBN b3VzdGFmYSA8YXNhYWQyQGdtYWlsLmNvbT7CwVwEEAECABAFAlGERIcJEJNQ R5iRS/V2AADgKQ/+KElMrKbcPDkHvbqQO28hEWeS8Fj4y/Td/aqp5ETC3BMG X5rxRlW5tVBKXIuUlEFugv9lUelLZIhrUFXpiJ6t9E/znS1d5gs2F+614x7S /RlUvEufTChmzES3cSzL/1hDO7NMz0xU7p5WAMxT7WrSvUEUV2j64o+eJApS I2Np/ZyBPx6OiDv4ttezJrU61R4tVYkVGt1ieCtaVAiTb8E7GhRXR0qihNM+ HNtutY5bieS52vZcLmHGl0GA8j6POMihCdiU3CscCrbJXc1PrdEszowme5xI vj84BgPZOtTJTWJhSdx422PwjgbBURVirTKunF/mXf+3X2GRvzXXzgx/o6Nk RXGGPZOQy39FQg5mNufpAljbfFeefgkcWE+iYLs8+wk7/4vcSoncvvdZxOCg onaK93S1GcFyAhKjVBKI1S0+MkFY1hpiGxU4kvnkDnrVDFCwqv+lp7mPAx3v 09vHBJ7rNY/WNivPurLpgrxA2s6ASY3Ocu05VRJ+A/a8IRJLGS87pLXDiH2O 4lrTvy9QQABgtAyQTmWVzx1kyso8K45odJQuAD6AKgirQK+Npc133ELugYaX nM8FbQKqMypxDYgd+Vgb2pFKibBpVI4h1CmOKHVYpZx0IaWI6fLkC0nt5mv3 6IcBsWjYc17ywKZc+4c8cD+AhBcGgbFMJstm3Dk= =mI3V -----END PGP PUBLIC KEY BLOCK----- On May 13, 2013 1:50 PM, "Alex Lake" wrote: > Is there a clever way in which one can use git/github to tell me (in > human readable terms!) all the changes between version A and version B > of Freeswitch? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/b7eb7f56/attachment.html From schoch+freeswitch.org at xwin32.com Mon May 13 22:07:37 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Mon, 13 May 2013 11:07:37 -0700 Subject: [Freeswitch-users] TCP ACK Ping In-Reply-To: <51900FA2.9020509@5ninesolutions.com> References: <51900FA2.9020509@5ninesolutions.com> Message-ID: On Sun, May 12, 2013 at 2:54 PM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > The phones periodically send a TCP ACK ping to > Freeswitch which goes unanswered and then phone then tears down the TCP > connection. I'm not at all a TCP expert but should Freeswitch be > responding to this unsolicited ACK? I've been working with TCP for 25 years, and this is the first I have heard of ACK ping. A quick Google search told me that this technique involves sending a normal TCP ACK packet (which all TCP packets except the initial one have, by the way) to a random TCP port. The host will then respond with a RST packet. (Which means reset the connection because the ACK was not sent to an established connection.) However, a firewall may filter and discard these random ACK packets. Since they're not part of "normal" TCP, no one cares. In answer to your question, this is in the TCP layer much deeper than Freeswitch. There is nothing Freeswitch can do at this level. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/85a61601/attachment.html From ben at langfeld.co.uk Mon May 13 22:15:46 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 13 May 2013 15:15:46 -0300 Subject: [Freeswitch-users] Changes between versions In-Reply-To: References: <51912132.4080407@digitalmail.com> Message-ID: Unfortunately you can't use github's swanky compare view because whoever it is that's syncing the repo there is not also pushing tags :( Can this be fixed? Regards, Ben Langfeld On 13 May 2013 15:05, BookBag wrote: > > -----BEGIN PGP PUBLIC KEY BLOCK----- > Version: OpenPGP.js v.1.20130228 > Comment: http://openpgpjs.org > > xsFNBFGERFIBEACeX3R6n+D3jK8KluJj0fTZOazMAh5JRdBDjeqI139T+uln > SzVeVXcgb7Gs3Ag5Ugo9S+nqPWGmuUOC1w39LMpDixYu7G5Cy1CG4wtw3ALR > kixYvqNPEHA4vwBq1jSLcRWeXNgrd97aWGltYXYjI7ZorPrWefLGJp7/24Sf > iUyCtD37EXYGX4a34lmbjH20qhZCAONlC9wyXdvJLkm4Nu4yQC4DujC7wWwQ > Wj8AFm6IZiqVuJTs8B9bdLrK3a1KgpZggSq1zWHppljplQseWsKz/4aXbOtY > v2BpCCyFBscFJhaoFpATRpY1RT3PT6Jf4JJiBK0UWp9akKycp5D8/+cetO/4 > bekXQP7dXtooV9+e1iwCccg+/cZaGUKChry+C1wMbe/8pW3KgaYq2SCIUFjh > +Q07CVEkwvkNHgrUy48FI9eQQ7cKnaIF+3z8Rh2OkxczB95ym1saN62WqAkK > bF1wpDGxFSkTiT60k6ioFqE2Lo+F+2JFA3Acpyp2gLsHnpT4/pQ5viSp+3ph > 5i7ZosoWli+JHNFGJL6P3RU+QoUJQ/carcBhGSx4xhgo9gNjQU8ImiJbbWfp > aMX9YmuKvIp4ZsslkDhWWUpkMgBTb97SfKDvwZ8zaQQAC2KKFsH9fVmPFAvu > J1CJ858NCU7Ys04R91kfK5EdBkA+z5vYmdlwVQARAQABzSRNb2hhbW1lZCBN > b3VzdGFmYSA8YXNhYWQyQGdtYWlsLmNvbT7CwVwEEAECABAFAlGERIcJEJNQ > R5iRS/V2AADgKQ/+KElMrKbcPDkHvbqQO28hEWeS8Fj4y/Td/aqp5ETC3BMG > X5rxRlW5tVBKXIuUlEFugv9lUelLZIhrUFXpiJ6t9E/znS1d5gs2F+614x7S > /RlUvEufTChmzES3cSzL/1hDO7NMz0xU7p5WAMxT7WrSvUEUV2j64o+eJApS > I2Np/ZyBPx6OiDv4ttezJrU61R4tVYkVGt1ieCtaVAiTb8E7GhRXR0qihNM+ > HNtutY5bieS52vZcLmHGl0GA8j6POMihCdiU3CscCrbJXc1PrdEszowme5xI > vj84BgPZOtTJTWJhSdx422PwjgbBURVirTKunF/mXf+3X2GRvzXXzgx/o6Nk > RXGGPZOQy39FQg5mNufpAljbfFeefgkcWE+iYLs8+wk7/4vcSoncvvdZxOCg > onaK93S1GcFyAhKjVBKI1S0+MkFY1hpiGxU4kvnkDnrVDFCwqv+lp7mPAx3v > 09vHBJ7rNY/WNivPurLpgrxA2s6ASY3Ocu05VRJ+A/a8IRJLGS87pLXDiH2O > 4lrTvy9QQABgtAyQTmWVzx1kyso8K45odJQuAD6AKgirQK+Npc133ELugYaX > nM8FbQKqMypxDYgd+Vgb2pFKibBpVI4h1CmOKHVYpZx0IaWI6fLkC0nt5mv3 > 6IcBsWjYc17ywKZc+4c8cD+AhBcGgbFMJstm3Dk= > =mI3V > -----END PGP PUBLIC KEY BLOCK----- > On May 13, 2013 1:50 PM, "Alex Lake" wrote: > >> Is there a clever way in which one can use git/github to tell me (in >> human readable terms!) all the changes between version A and version B >> of Freeswitch? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/d8d49a3c/attachment-0001.html From krice at freeswitch.org Mon May 13 22:27:28 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 13:27:28 -0500 Subject: [Freeswitch-users] Changes between versions In-Reply-To: Message-ID: http://git.freeswitch.org/ On 5/13/13 1:15 PM, "Ben Langfeld" wrote: > Unfortunately you can't use github's swanky compare view because whoever it is > that's syncing the repo there is not also pushing tags :( Can this be fixed? > > Regards, > Ben Langfeld > > > On 13 May 2013 15:05, BookBag wrote: >> >> >> >> -----BEGIN PGP PUBLIC KEY BLOCK----- >> Version: OpenPGP.js v.1.20130228 >> Comment: http://openpgpjs.org >> >> xsFNBFGERFIBEACeX3R6n+D3jK8KluJj0fTZOazMAh5JRdBDjeqI139T+uln >> SzVeVXcgb7Gs3Ag5Ugo9S+nqPWGmuUOC1w39LMpDixYu7G5Cy1CG4wtw3ALR >> kixYvqNPEHA4vwBq1jSLcRWeXNgrd97aWGltYXYjI7ZorPrWefLGJp7/24Sf >> iUyCtD37EXYGX4a34lmbjH20qhZCAONlC9wyXdvJLkm4Nu4yQC4DujC7wWwQ >> Wj8AFm6IZiqVuJTs8B9bdLrK3a1KgpZggSq1zWHppljplQseWsKz/4aXbOtY >> v2BpCCyFBscFJhaoFpATRpY1RT3PT6Jf4JJiBK0UWp9akKycp5D8/+cetO/4 >> bekXQP7dXtooV9+e1iwCccg+/cZaGUKChry+C1wMbe/8pW3KgaYq2SCIUFjh >> +Q07CVEkwvkNHgrUy48FI9eQQ7cKnaIF+3z8Rh2OkxczB95ym1saN62WqAkK >> bF1wpDGxFSkTiT60k6ioFqE2Lo+F+2JFA3Acpyp2gLsHnpT4/pQ5viSp+3ph >> 5i7ZosoWli+JHNFGJL6P3RU+QoUJQ/carcBhGSx4xhgo9gNjQU8ImiJbbWfp >> aMX9YmuKvIp4ZsslkDhWWUpkMgBTb97SfKDvwZ8zaQQAC2KKFsH9fVmPFAvu >> J1CJ858NCU7Ys04R91kfK5EdBkA+z5vYmdlwVQARAQABzSRNb2hhbW1lZCBN >> b3VzdGFmYSA8YXNhYWQyQGdtYWlsLmNvbT7CwVwEEAECABAFAlGERIcJEJNQ >> R5iRS/V2AADgKQ/+KElMrKbcPDkHvbqQO28hEWeS8Fj4y/Td/aqp5ETC3BMG >> X5rxRlW5tVBKXIuUlEFugv9lUelLZIhrUFXpiJ6t9E/znS1d5gs2F+614x7S >> /RlUvEufTChmzES3cSzL/1hDO7NMz0xU7p5WAMxT7WrSvUEUV2j64o+eJApS >> I2Np/ZyBPx6OiDv4ttezJrU61R4tVYkVGt1ieCtaVAiTb8E7GhRXR0qihNM+ >> HNtutY5bieS52vZcLmHGl0GA8j6POMihCdiU3CscCrbJXc1PrdEszowme5xI >> vj84BgPZOtTJTWJhSdx422PwjgbBURVirTKunF/mXf+3X2GRvzXXzgx/o6Nk >> RXGGPZOQy39FQg5mNufpAljbfFeefgkcWE+iYLs8+wk7/4vcSoncvvdZxOCg >> onaK93S1GcFyAhKjVBKI1S0+MkFY1hpiGxU4kvnkDnrVDFCwqv+lp7mPAx3v >> 09vHBJ7rNY/WNivPurLpgrxA2s6ASY3Ocu05VRJ+A/a8IRJLGS87pLXDiH2O >> 4lrTvy9QQABgtAyQTmWVzx1kyso8K45odJQuAD6AKgirQK+Npc133ELugYaX >> nM8FbQKqMypxDYgd+Vgb2pFKibBpVI4h1CmOKHVYpZx0IaWI6fLkC0nt5mv3 >> 6IcBsWjYc17ywKZc+4c8cD+AhBcGgbFMJstm3Dk= >> =mI3V >> -----END PGP PUBLIC KEY BLOCK----- >> >> On May 13, 2013 1:50 PM, "Alex Lake" wrote: >>> Is there a clever way in which one can use git/github to tell me (in >>> human readable terms!) all the changes between version A and version B >>> of Freeswitch? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/ab72a0b2/attachment.html From spencer at 5ninesolutions.com Mon May 13 23:37:37 2013 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 13 May 2013 15:37:37 -0400 Subject: [Freeswitch-users] TCP ACK Ping In-Reply-To: References: <51900FA2.9020509@5ninesolutions.com> Message-ID: <51914101.50504@5ninesolutions.com> Understood, I though it was quite strange to see these unsolicited ACKs from the Polycoms followed by a RST. Thanks, Spencer On 05/13/2013 02:07 PM, Steven Schoch wrote: > On Sun, May 12, 2013 at 2:54 PM, Spencer Thomason > > wrote: > > The phones periodically send a TCP ACK ping to > Freeswitch which goes unanswered and then phone then tears down > the TCP > connection. I'm not at all a TCP expert but should Freeswitch be > responding to this unsolicited ACK? > > > I've been working with TCP for 25 years, and this is the first I have > heard of ACK ping. A quick Google search told me that this technique > involves sending a normal TCP ACK packet (which all TCP packets except > the initial one have, by the way) to a random TCP port. The host will > then respond with a RST packet. (Which means reset the connection > because the ACK was not sent to an established connection.) > > However, a firewall may filter and discard these random ACK packets. > Since they're not part of "normal" TCP, no one cares. > > In answer to your question, this is in the TCP layer much deeper than > Freeswitch. There is nothing Freeswitch can do at this level. > > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/84df56fa/attachment.html From intralanman at freeswitch.org Mon May 13 23:38:44 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Mon, 13 May 2013 15:38:44 -0400 Subject: [Freeswitch-users] Changes between versions In-Reply-To: References: Message-ID: <51914144.5070901@freeswitch.org> On 13-05-13 01:58 PM, Ken Rice wrote: > Git log tag..tag > I think > i think it's git diff tag..tag -Ray From fs-list at communicatefreely.net Mon May 13 23:47:56 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 13 May 2013 15:47:56 -0400 Subject: [Freeswitch-users] Sofia profile for dual-stack IPv4/v6 Message-ID: <5191436C.3040705@communicatefreely.net> Hello, I'm implementing FreeSWITCH in a dual-stack environment, now that we have found some endpoints that can do IPv6 properly. I have two profiles - internal and internal-ipv6. I can make calls from either, but in order to place calls so that a user could be reachable on either profile, I had to use sofia_contact(*/${dialed_user}@${dialed_domain}) This seems to work just fine, but from what I understand, it will try ALL profiles. Is there a way to have it just try specific profiles? I tried it without the */ (just user at domain) and it only came back if it was in the internal profile. This could be because I can only alias a domain to one profile at a time, and I have aliased them all to internal. For reference, here's my profile arrangement: external - trunking only, no registrations, all static IP and gateways external-alt - same as above, on an alternate IP address public - no registrations, a restricted profile just for incoming sip calls (URI calling, enum, etc.) public-ipv6 - same as above, but IPv6 fax - all our fax ATAs register to this. They all have the same domain internal - all our endpoints register here, all customer domains are aliased to this profile internal-ipv6 - IPv6 endpoints register here. We can't alias the domains, as they are already aliased above. The profile is otherwise the same, except that it is bound to an IPv6 interface, has NAT turned off, and will soon allow bypass media. Has anyone else run FreeSWITCH in a dual-stack environment? Are there any potential negative consequences to using the */ option in sofia_contact? Thanks!! -Tim From ben at langfeld.co.uk Mon May 13 23:54:06 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 13 May 2013 16:54:06 -0300 Subject: [Freeswitch-users] Changes between versions In-Reply-To: References: Message-ID: That's great, Ken, but I'm not seeing anything there quite as convenient as https://github.com/rails/rails/compare/v3.1.0...v3.2.0, for example :) Regards, Ben Langfeld On 13 May 2013 15:27, Ken Rice wrote: > http://git.freeswitch.org/ > > > > On 5/13/13 1:15 PM, "Ben Langfeld" wrote: > > Unfortunately you can't use github's swanky compare view because whoever > it is that's syncing the repo there is not also pushing tags :( Can this be > fixed? > > Regards, > Ben Langfeld > > > On 13 May 2013 15:05, BookBag wrote: > > > > > -----BEGIN PGP PUBLIC KEY BLOCK----- > Version: OpenPGP.js v.1.20130228 > Comment: http://openpgpjs.org > > xsFNBFGERFIBEACeX3R6n+D3jK8KluJj0fTZOazMAh5JRdBDjeqI139T+uln > SzVeVXcgb7Gs3Ag5Ugo9S+nqPWGmuUOC1w39LMpDixYu7G5Cy1CG4wtw3ALR > kixYvqNPEHA4vwBq1jSLcRWeXNgrd97aWGltYXYjI7ZorPrWefLGJp7/24Sf > iUyCtD37EXYGX4a34lmbjH20qhZCAONlC9wyXdvJLkm4Nu4yQC4DujC7wWwQ > Wj8AFm6IZiqVuJTs8B9bdLrK3a1KgpZggSq1zWHppljplQseWsKz/4aXbOtY > v2BpCCyFBscFJhaoFpATRpY1RT3PT6Jf4JJiBK0UWp9akKycp5D8/+cetO/4 > bekXQP7dXtooV9+e1iwCccg+/cZaGUKChry+C1wMbe/8pW3KgaYq2SCIUFjh > +Q07CVEkwvkNHgrUy48FI9eQQ7cKnaIF+3z8Rh2OkxczB95ym1saN62WqAkK > bF1wpDGxFSkTiT60k6ioFqE2Lo+F+2JFA3Acpyp2gLsHnpT4/pQ5viSp+3ph > 5i7ZosoWli+JHNFGJL6P3RU+QoUJQ/carcBhGSx4xhgo9gNjQU8ImiJbbWfp > aMX9YmuKvIp4ZsslkDhWWUpkMgBTb97SfKDvwZ8zaQQAC2KKFsH9fVmPFAvu > J1CJ858NCU7Ys04R91kfK5EdBkA+z5vYmdlwVQARAQABzSRNb2hhbW1lZCBN > b3VzdGFmYSA8YXNhYWQyQGdtYWlsLmNvbT7CwVwEEAECABAFAlGERIcJEJNQ > R5iRS/V2AADgKQ/+KElMrKbcPDkHvbqQO28hEWeS8Fj4y/Td/aqp5ETC3BMG > X5rxRlW5tVBKXIuUlEFugv9lUelLZIhrUFXpiJ6t9E/znS1d5gs2F+614x7S > /RlUvEufTChmzES3cSzL/1hDO7NMz0xU7p5WAMxT7WrSvUEUV2j64o+eJApS > I2Np/ZyBPx6OiDv4ttezJrU61R4tVYkVGt1ieCtaVAiTb8E7GhRXR0qihNM+ > HNtutY5bieS52vZcLmHGl0GA8j6POMihCdiU3CscCrbJXc1PrdEszowme5xI > vj84BgPZOtTJTWJhSdx422PwjgbBURVirTKunF/mXf+3X2GRvzXXzgx/o6Nk > RXGGPZOQy39FQg5mNufpAljbfFeefgkcWE+iYLs8+wk7/4vcSoncvvdZxOCg > onaK93S1GcFyAhKjVBKI1S0+MkFY1hpiGxU4kvnkDnrVDFCwqv+lp7mPAx3v > 09vHBJ7rNY/WNivPurLpgrxA2s6ASY3Ocu05VRJ+A/a8IRJLGS87pLXDiH2O > 4lrTvy9QQABgtAyQTmWVzx1kyso8K45odJQuAD6AKgirQK+Npc133ELugYaX > nM8FbQKqMypxDYgd+Vgb2pFKibBpVI4h1CmOKHVYpZx0IaWI6fLkC0nt5mv3 > 6IcBsWjYc17ywKZc+4c8cD+AhBcGgbFMJstm3Dk= > =mI3V > -----END PGP PUBLIC KEY BLOCK----- > > On May 13, 2013 1:50 PM, "Alex Lake" wrote: > > Is there a clever way in which one can use git/github to tell me (in > human readable terms!) all the changes between version A and version B > of Freeswitch? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/fa2318f2/attachment-0001.html From william.suffill at gmail.com Tue May 14 00:15:34 2013 From: william.suffill at gmail.com (William Suffill) Date: Mon, 13 May 2013 16:15:34 -0400 Subject: [Freeswitch-users] Changes between versions In-Reply-To: References: Message-ID: Why not push the tags to GitHub as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/4d87cec1/attachment.html From john at telecube.com.au Mon May 13 21:16:11 2013 From: john at telecube.com.au (Telecube - John) Date: Tue, 14 May 2013 03:16:11 +1000 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: <5190FE16.6090502@telecube.com.au> Message-ID: <51911FDB.4060602@telecube.com.au> Yep, fail2ban is another option.. it's nice to have options.. ;-) -John On 14/05/13 2:28 AM, Steven Ayre wrote: > This is the purpose of the fail2ban system that was already mentioned. > > -Steve > > > > On 13 May 2013 15:52, Telecube - John > wrote: > > You can install a log watching program that looks out for specific > log entries and can take action. Eg; writing a firewall rule to > ban the ip address of the system trying to make unauthorised calls. > > 'swatch' is a good tool for this in linux systems. > > http://sourceforge.net/projects/swatch/ > > John > > On 14/05/13 12:35 AM, Ken Rice wrote: >> Ok look at the 3rd column... This is someone trying to figure out >> how to get calls thru your public context with various prefix >> codexs... Notices how all the calls end with 972598371070 <-- >> that's the intl destination umber probably a phone in gaza >> (definitely in Isreali controlled areas, but I think I looked up >> this number or similar number in the past) >> >> To protect yourself, only allow calls to local extentsions from >> the public context... >> >> >> On 5/13/13 2:49 AM, "Steven Ayre" > > wrote: >> >> The call duration is 0 seconds so no they didn't get in. Any >> accepted call attempt will generate a CDR even if it was not >> answered. >> >> They're coming in on the 'public' context which'll mean an >> unauthenticated call on the external profile. That would be >> the context allowing you to receive calls from strangers. >> Providing that context doesn't allow them to dial out via >> providers that charge you this isn't an issue. >> >> -Steve >> >> >> >> >> >> >> On 13 May 2013 07:17, Bala Murugan Mahendran > > wrote: >> >> I have lot of entries like 1000,1001,1002 and so on, Is >> someone trying to get inside? I believe they didn't in >> yet but how come we have cdr log like this? >> >> "1004","1004","012972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","013972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","014972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","010972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0061972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0041972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","000972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","006972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","002972598371070","public","2013-05-12 >> 07:30:10","","2013-05-12 >> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","810972598371070","public","2013-05-12 >> 07:29:49","","2013-05-12 >> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" >> >> >> Thanks, >> Bala >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> ------------------------------------------------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/5b3bc652/attachment-0001.html From tru083 at yahoo.com Mon May 13 23:31:23 2013 From: tru083 at yahoo.com (D D) Date: Mon, 13 May 2013 12:31:23 -0700 (PDT) Subject: [Freeswitch-users] eavesdrop not sending video In-Reply-To: References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> Message-ID: <1368473483.38110.YahooMailNeo@web120701.mail.ne1.yahoo.com> The wiki page http://wiki.freeswitch.org/wiki/Video-recording says that video eavesdrop should work. ? Which source file should I check in order to verify the implementation? ________________________________ From: Ken Rice To: FreeSWITCH Users Help Sent: Monday, May 13, 2013 11:19 AM Subject: Re: [Freeswitch-users] eavesdrop not sending video Re: [Freeswitch-users] eavesdrop not sending video Eavesdrop probably does not support sending the video stream as well... If I recall correctly, it only taps the audio stream... ?Possible solutions, open a jira and offer a bounty for someone to look at adding support for video On 5/13/13 10:54 AM, "D D" wrote: Hi, > >I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. > >I have followed the directions described at http://wiki.freeswitch.org/wiki/Video-recording >(including a rebuild after doing the git checkout of video-media-bug), and I can successfully >use record_fsv to record the video to a file. > >Any ideas why the eavesdrop does not send the video? > >Thanks, >David > > >>________________________________ >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130513/33e95a01/attachment.html From krice at freeswitch.org Tue May 14 00:47:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 15:47:15 -0500 Subject: [Freeswitch-users] eavesdrop not sending video In-Reply-To: <1368473483.38110.YahooMailNeo@web120701.mail.ne1.yahoo.com> Message-ID: Actually I believe the wiki is wrong here, and I confirmed with Anthony, eavesdrop does not work on video... The is probably in mod_commands but I am not sure On 5/13/13 2:31 PM, "D D" wrote: > The wiki page http://wiki.freeswitch.org/wiki/Video-recording says that video > eavesdrop should work. > > Which source file should I check in order to verify the implementation? > > > > > > > > > From: Ken Rice > To: FreeSWITCH Users Help > Sent: Monday, May 13, 2013 11:19 AM > Subject: Re: [Freeswitch-users] eavesdrop not sending video > > > > Re: [Freeswitch-users] eavesdrop not sending video > Eavesdrop probably does not support sending the video stream as well... If I > recall correctly, it only taps the audio stream... Possible solutions, open a > jira and offer a bounty for someone to look at adding support for video > > On 5/13/13 10:54 AM, "D D" wrote: > >> Hi, >> >> I tried to use eavesdrop to capture the video of a parked call, but I only >> receive the audio. >> >> I have followed the directions described at >> http://wiki.freeswitch.org/wiki/Video-recording >> (including a rebuild after doing the git checkout of video-media-bug), and I >> can successfully >> use record_fsv to record the video to a file. >> >> Any ideas why the eavesdrop does not send the video? >> >> Thanks, >> David >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/aeaa15b3/attachment.html From krice at freeswitch.org Tue May 14 00:52:38 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 15:52:38 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... Message-ID: Hey Guys, Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 and I said packages were coming. We now have both YUM and Debian Repo?s for you guys to install from... For Debian check out http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packa ges For Centos check out http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_In stallation These will get you the stable branch for now... TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for it -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/c9722bb1/attachment.html From jpyle at fidelityvoice.com Tue May 14 00:55:39 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 13 May 2013 16:55:39 -0400 Subject: [Freeswitch-users] pass "called" RPID from B-leg to A-leg Message-ID: <5191534B.9060706@fidelityvoice.com> Hello, Searching the archives I've seen this question asked a few times...but never answered. Or, at least not in a way that my feeble neurons can process. A-leg is bridged to B-leg with: The B-leg responds with a Remote-Party-Id header in a 180 Ringing. How can I pass this information into the 180 Ringing that gets send back to the A-leg? It seems I may need to set sip_callee_id_number/name at some point, but I'm not sure how to get at the B-leg data to set those variables. - Jeff From steveayre at gmail.com Tue May 14 01:18:41 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 22:18:41 +0100 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: Great news! Any idea when Wheezy packages will be added? -Steve On 13 May 2013 21:52, Ken Rice wrote: > Hey Guys, > > Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 > and I said packages were coming. > > We now have both YUM and Debian Repo?s for you guys to install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for > it > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/c96bd560/attachment.html From krice at freeswitch.org Tue May 14 01:25:19 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 16:25:19 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: Message-ID: There are packages there for Squeeze, Wheezy and Sid... Jesse packages are in the early stages On 5/13/13 4:18 PM, "Steven Ayre" wrote: > Great news! Any idea when Wheezy packages will be added? > > -Steve > > > On 13 May 2013 21:52, Ken Rice wrote: >> Hey Guys, >> >> Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 and >> I said packages were coming. >> >> We now have both YUM and Debian Repo?s for you guys to install from... >> >> For Debian check out >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packag >> es >> For Centos check out >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Ins >> tallation >> >> These will get you the stable branch for now... >> >> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for it >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/6a1f749f/attachment.html From steveayre at gmail.com Tue May 14 01:30:33 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 13 May 2013 22:30:33 +0100 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: Ah ok. I only saw squeeze and sid just now, but I see wheezy there now. Perhaps it was an out of date mirror? :) On 13 May 2013 22:25, Ken Rice wrote: > There are packages there for Squeeze, Wheezy and Sid... Jesse packages > are in the early stages > > > > On 5/13/13 4:18 PM, "Steven Ayre" wrote: > > Great news! Any idea when Wheezy packages will be added? > > -Steve > > > On 13 May 2013 21:52, Ken Rice wrote: > > Hey Guys, > > Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 > and I said packages were coming. > > We now have both YUM and Debian Repo?s for you guys to install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for > it > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/65b7b70b/attachment.html From krice at freeswitch.org Tue May 14 01:35:05 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 16:35:05 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: Message-ID: Depends on which mirror on the CDN you hit... They can be slow at updating sometimes On 5/13/13 4:30 PM, "Steven Ayre" wrote: > Ah ok. I only saw squeeze and sid just now, but I see wheezy there now. > Perhaps it was an out of date mirror? :) > > > > On 13 May 2013 22:25, Ken Rice wrote: >> There are packages there for Squeeze, Wheezy and Sid... Jesse packages are in >> the early stages >> >> >> >> On 5/13/13 4:18 PM, "Steven Ayre" > > wrote: >> >>> Great news! Any idea when Wheezy packages will be added? >>> >>> -Steve >>> >>> >>> On 13 May 2013 21:52, Ken Rice >> > wrote: >>>> Hey Guys, >>>> >>>> Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 >>>> and I said packages were coming. >>>> >>>> We now have both YUM and Debian Repo?s for you guys to install from... >>>> >>>> For Debian check out >>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_pack >>>> ages >>>> For Centos check out >>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_I >>>> nstallation >>>> >>>> These will get you the stable branch for now... >>>> >>>> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for >>>> it >>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/eaea6705/attachment.html From jeff at askcornerstone.net Tue May 14 01:38:08 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Mon, 13 May 2013 21:38:08 +0000 Subject: [Freeswitch-users] Recommendations for Trunks from Hawaii? In-Reply-To: References: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CDC2C@mail2.firstdataworks.net> Looks like they don't have any Hawaii area code DIDs... :( Jeff Bernhardt Systems Administrator Cornerstone Consulting 808.440.2900 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: Monday, May 13, 2013 2:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Recommendations for Trunks from Hawaii? Flowroute is in Las Vegas and I think Los Angeles. Might wantvto check them out: http://flowroute.com -BDF On May 13, 2013 4:11 AM, "Jeff Bernhardt" > wrote: Not Freeswitch specific, but... We have a couple Vitelity DIDs we use for testing, but the latency on them is fairly bad. We figure this is just because of physical distance... we're in Hawaii and Vitelity is I believe in Denver (at least that's the ip we connect to). Does anyone have recommendations for any good trunks that might be located somewhere closer (California)? There are plenty of companies here and there, but what are people using that might meet our needs? Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130513/9fb21e6a/attachment-0001.html From jpyle at fidelityvoice.com Tue May 14 02:11:37 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 13 May 2013 18:11:37 -0400 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: <51916519.3030203@fidelityvoice.com> Same situation here? apt-get complaining: Package freeswitch-mod-ilbc is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from another source Package freeswitch-mod-siren is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from another source E: Package 'freeswitch-mod-siren' has no installation candidate E: Package 'freeswitch-mod-ilbc' has no installation candidate Just give it a shot later? - Jeff Ken Rice wrote: > Depends on which mirror on the CDN you hit... They can be slow at > updating sometimes > > On 5/13/13 4:30 PM, "Steven Ayre" wrote: > > Ah ok. I only saw squeeze and sid just now, but I see wheezy there > now. Perhaps it was an out of date mirror? :) > > > > On 13 May 2013 22:25, Ken Rice wrote: > > There are packages there for Squeeze, Wheezy and Sid... Jesse > packages are in the early stages > > > > On 5/13/13 4:18 PM, "Steven Ayre" > wrote: > > Great news! Any idea when Wheezy packages will be added? > > -Steve > > > On 13 May 2013 21:52, Ken Rice > wrote: > > Hey Guys, > > Just a quick heads up, As you know last week we > released FreeSWITCH 1.2.9 and I said packages were > coming. > > We now have both YUM and Debian Repo?s for you guys to > install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the > deb?s and rpm?s for it > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/c76cdd44/attachment.html From krice at freeswitch.org Tue May 14 02:17:14 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 17:17:14 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <51916519.3030203@fidelityvoice.com> Message-ID: I don?t know if we are packaging those... We should be tho can you open a ticket on that? On 5/13/13 5:11 PM, "Jeff Pyle" wrote: > Same situation here?? apt-get complaining: > > Package freeswitch-mod-ilbc is not available, but is referred to by another > package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > Package freeswitch-mod-siren is not available, but is referred to by another > package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > E: Package 'freeswitch-mod-siren' has no installation candidate > E: Package 'freeswitch-mod-ilbc' has no installation candidate > > Just give it a shot later? > > > - Jeff > > > Ken Rice wrote: >> Re: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... Depends on >> which mirror on the CDN you hit... They can be slow at updating sometimes >> >> On 5/13/13 4:30 PM, "Steven Ayre" wrote: >> >> >>> Ah ok. I only saw squeeze and sid just now, but I see wheezy there now. >>> Perhaps it was an out of date mirror? :) >>> >>> >>> >>> On 13 May 2013 22:25, Ken Rice wrote: >>>> There are packages there for Squeeze, Wheezy and Sid... Jesse packages are >>>> in the early stages >>>> >>>> >>>> >>>> On 5/13/13 4:18 PM, "Steven Ayre" >>> > wrote: >>>> >>>>> Great news! Any idea when Wheezy packages will be added? >>>>> >>>>> -Steve >>>>> >>>>> >>>>> On 13 May 2013 21:52, Ken Rice >>>> > wrote: >>>>>> Hey Guys, >>>>>> >>>>>> Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 >>>>>> and I said packages were coming. >>>>>> >>>>>> We now have both YUM and Debian Repo?s for you guys to install from... >>>>>> >>>>>> For Debian check out >>>>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_pa >>>>>> ckages >>>>>> For Centos check out >>>>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based >>>>>> _Installation >>>>>> >>>>>> These will get you the stable branch for now... >>>>>> >>>>>> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for >>>>>> it >>>>>> >> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/8ed0345d/attachment.html From william.king at quentustech.com Tue May 14 02:20:25 2013 From: william.king at quentustech.com (William King) Date: Mon, 13 May 2013 15:20:25 -0700 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <51916519.3030203@fidelityvoice.com> References: <51916519.3030203@fidelityvoice.com> Message-ID: <51916729.3020501@quentustech.com> Which meta package did you try to install? 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 05/13/2013 03:11 PM, Jeff Pyle wrote: > Same situation here? apt-get complaining: > > Package freeswitch-mod-ilbc is not available, but is referred to by > another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > Package freeswitch-mod-siren is not available, but is referred to by > another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > E: Package 'freeswitch-mod-siren' has no installation candidate > E: Package 'freeswitch-mod-ilbc' has no installation candidate > > Just give it a shot later? > > > - Jeff > > > Ken Rice wrote: >> Depends on which mirror on the CDN you hit... They can be slow at >> updating sometimes >> >> On 5/13/13 4:30 PM, "Steven Ayre" wrote: >> >> Ah ok. I only saw squeeze and sid just now, but I see wheezy there >> now. Perhaps it was an out of date mirror? :) >> >> >> >> On 13 May 2013 22:25, Ken Rice wrote: >> >> There are packages there for Squeeze, Wheezy and Sid... Jesse >> packages are in the early stages >> >> >> >> On 5/13/13 4:18 PM, "Steven Ayre" > > wrote: >> >> Great news! Any idea when Wheezy packages will be added? >> >> -Steve >> >> >> On 13 May 2013 21:52, Ken Rice > > wrote: >> >> Hey Guys, >> >> Just a quick heads up, As you know last week we >> released FreeSWITCH 1.2.9 and I said packages were >> coming. >> >> We now have both YUM and Debian Repo?s for you guys to >> install from... >> >> For Debian check out >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages >> For Centos check out >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >> >> These will get you the stable branch for now... >> >> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the >> deb?s and rpm?s for it >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From jeff at askcornerstone.net Tue May 14 02:04:49 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Mon, 13 May 2013 22:04:49 +0000 Subject: [Freeswitch-users] Recommendations for Trunks from Hawaii? In-Reply-To: <5190E2B8.5060306@millican.us> References: <8A9716A5B256904FB1F07C050F9CCCCB020CD767@mail2.firstdataworks.net> <5190E2B8.5060306@millican.us> Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CDC97@mail2.firstdataworks.net> Yeah, actually, ping to the Vitelity server we connect to is about 95ms as well, but I wonder if they route SIP traffic elsewhere in some confounded way to get to Hawaii. Maybe I should actually just email Vitelity and ask! No satellite link for Hawaiian providers. There's a network of submarine cables that connects us to the rest of the world. Check this out: http://www.submarinecablemap.com/ Jeff From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of john at millican.us Sent: Monday, May 13, 2013 2:55 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Recommendations for Trunks from Hawaii? On 5/13/2013 8:00 AM, Brian Foster wrote: Flowroute is in Las Vegas and I think Los Angeles. Might wantvto check them out: http://flowroute.com -BDF On May 13, 2013 4:11 AM, "Jeff Bernhardt" > wrote: Not Freeswitch specific, but... We have a couple Vitelity DIDs we use for testing, but the latency on them is fairly bad. We figure this is just because of physical distance... we're in Hawaii and Vitelity is I believe in Denver (at least that's the ip we connect to). Does anyone have recommendations for any good trunks that might be located somewhere closer (California)? There are plenty of companies here and there, but what are people using that might meet our needs? Thanks. Speaking based only on "assumptions" here but I doubt if there will be much difference between Hawaii and a SIP provider in LA or one in Denver. I have had servers of my own in New Hampshire, Virginia, and LA and the max ping times over public Internet were typically around 95ms. I would bet that the majority of the latency you are seeing is due to the link between Hawaii and the mainland. Could it be a satellite link that your Internet provider is using? Have you asked them about the connectivity they use to get to mainland destinations? JohnM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130513/757f0938/attachment-0001.html From mishehu at freeswitch.org Tue May 14 03:33:51 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Mon, 13 May 2013 18:33:51 -0500 Subject: [Freeswitch-users] Dynamic external profile proxies In-Reply-To: References: Message-ID: <5191785F.50302@freeswitch.org> SIP is the signalling, and RTP is the media stream. For load-balancing of SIP calls to different servers, you're probably better served by OpenSIPS or a similar project (i.e. kamailio). FreeSWITCH can readily interoperate with OpenSIPS and Kamailio. http://www.opensips.org http://www.kamailio.org -Yossi On 05/13/2013 10:40 AM, Oleg Stolyar wrote: > I am very new to FreeSWITCH and would like to use it as a SIP load > balancer to other (media) FreeSWITCH instances. I was thinking of > using the external SIP profile and bridging incoming calls to the > media instances of FreeSWITCH using some rotation mechanism (round > robin, least busy, etc) > > My question is this: Would I need to have a separate SIP profile for > each media FreeSWITCH server or can I define a profile in such a way > that I can dynamically specify the proxy for each call? > > Thank you > /Oleg/ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/dcd4cc71/attachment.html From mishehu at freeswitch.org Tue May 14 03:39:06 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Mon, 13 May 2013 18:39:06 -0500 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: References: Message-ID: <5191799A.1010000@freeswitch.org> I bet that DID is hosted by Jawal. Jawal is a Palestinian telco, but I think that 0598 is specifically cellular, so it's not limited to Gaza. I've seen these hitting other servers on the Interwebs as well... -Yossi On 05/13/2013 09:35 AM, Ken Rice wrote: > Re: [Freeswitch-users] Strange CDR entry Ok look at the 3rd column... > This is someone trying to figure out how to get calls thru your public > context with various prefix codexs... Notices how all the calls end > with 972598371070 <-- that's the intl destination umber probably a > phone in gaza (definitely in Isreali controlled areas, but I think I > looked up this number or similar number in the past) > > To protect yourself, only allow calls to local extentsions from the > public context... > > > On 5/13/13 2:49 AM, "Steven Ayre" wrote: > > The call duration is 0 seconds so no they didn't get in. Any > accepted call attempt will generate a CDR even if it was not answered. > > They're coming in on the 'public' context which'll mean an > unauthenticated call on the external profile. That would be the > context allowing you to receive calls from strangers. Providing > that context doesn't allow them to dial out via providers that > charge you this isn't an issue. > > -Steve > > > > > > > On 13 May 2013 07:17, Bala Murugan Mahendran wrote: > > I have lot of entries like 1000,1001,1002 and so on, Is > someone trying to get inside? I believe they didn't in yet but > how come we have cdr log like this? > > "1004","1004","012972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","013972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","014972598371070","public","2013-05-12 > 07:30:07","","2013-05-12 > 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","010972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0061972598371070","public","2013-05-12 > 07:30:08","","2013-05-12 > 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","0041972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" > "1004","1004","000972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" > "1004","1004","006972598371070","public","2013-05-12 > 07:30:09","","2013-05-12 > 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" > "1004","1004","002972598371070","public","2013-05-12 > 07:30:10","","2013-05-12 > 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" > "1004","1004","810972598371070","public","2013-05-12 > 07:29:49","","2013-05-12 > 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/6258d870/attachment.html From sirimmfs at gmail.com Tue May 14 04:31:58 2013 From: sirimmfs at gmail.com (Siri MM) Date: Tue, 14 May 2013 10:31:58 +1000 Subject: [Freeswitch-users] IP change detected -> no internal profile Message-ID: Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/8fddb457/attachment-0001.html From jpyle at fidelityvoice.com Tue May 14 04:39:04 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 13 May 2013 20:39:04 -0400 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: <519187A8.4080501@fidelityvoice.com> Done. FS-5415 . - Jeff Ken Rice wrote: > I don?t know if we are packaging those... We should be tho can you > open a ticket on that? > > > On 5/13/13 5:11 PM, "Jeff Pyle" wrote: > > Same situation here? apt-get complaining: > > Package freeswitch-mod-ilbc is not available, but is referred to > by another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > Package freeswitch-mod-siren is not available, but is referred to > by another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > E: Package 'freeswitch-mod-siren' has no installation candidate > E: Package 'freeswitch-mod-ilbc' has no installation candidate > > Just give it a shot later? > > > - Jeff > > > Ken Rice wrote: > > Re: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... > Depends on which mirror on the CDN you hit... They can be > slow at updating sometimes > > On 5/13/13 4:30 PM, "Steven Ayre" wrote: > > > Ah ok. I only saw squeeze and sid just now, but I see > wheezy there now. Perhaps it was an out of date mirror? :) > > > > On 13 May 2013 22:25, Ken Rice wrote: > > There are packages there for Squeeze, Wheezy and > Sid... Jesse packages are in the early stages > > > > On 5/13/13 4:18 PM, "Steven Ayre" > wrote: > > Great news! Any idea when Wheezy packages will be > added? > > -Steve > > > On 13 May 2013 21:52, Ken Rice > > wrote: > > Hey Guys, > > Just a quick heads up, As you know last week > we released FreeSWITCH 1.2.9 and I said > packages were coming. > > We now have both YUM and Debian Repo?s for you > guys to install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so > are the deb?s and rpm?s for it > > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/c1566748/attachment.html From jpyle at fidelityvoice.com Tue May 14 04:39:05 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Mon, 13 May 2013 20:39:05 -0400 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <51916729.3020501@quentustech.com> References: <51916519.3030203@fidelityvoice.com> <51916729.3020501@quentustech.com> Message-ID: <519187A9.2040102@fidelityvoice.com> My standard FS install on Debian includes the following packages: freeswitch freeswitch-init freeswitch-mod-bv freeswitch-mod-codec2 freeswitch-mod-commands freeswitch-mod-console freeswitch-mod-dialplan-xml freeswitch-mod-dptools freeswitch-mod-event-socket freeswitch-mod-g729 freeswitch-mod-hash freeswitch-mod-ilbc freeswitch-mod-logfile freeswitch-mod-opus freeswitch-mod-silk freeswitch-mod-siren freeswitch-mod-sofia freeswitch-mod-spandsp freeswitch-mod-speex freeswitch-mod-syslog freeswitch-mod-timerfd freeswitch-systemd freeswitch-sysvinit I noticed this thread mentioning 1.2.9 was available, so I decided to move from my custom repo from HEAD back to the one at files.freeswitch.org. And then I noticed these two packages weren't available. - Jeff William King wrote: > Which meta package did you try to install? > > 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 05/13/2013 03:11 PM, Jeff Pyle wrote: >> Same situation here? apt-get complaining: >> >> Package freeswitch-mod-ilbc is not available, but is referred to by >> another package. >> This may mean that the package is missing, has been obsoleted, or >> is only available from another source >> >> Package freeswitch-mod-siren is not available, but is referred to by >> another package. >> This may mean that the package is missing, has been obsoleted, or >> is only available from another source >> >> E: Package 'freeswitch-mod-siren' has no installation candidate >> E: Package 'freeswitch-mod-ilbc' has no installation candidate >> >> Just give it a shot later? >> >> >> - Jeff >> >> >> Ken Rice wrote: >>> Depends on which mirror on the CDN you hit... They can be slow at >>> updating sometimes >>> >>> On 5/13/13 4:30 PM, "Steven Ayre" wrote: >>> >>> Ah ok. I only saw squeeze and sid just now, but I see wheezy there >>> now. Perhaps it was an out of date mirror? :) >>> >>> >>> >>> On 13 May 2013 22:25, Ken Rice wrote: >>> >>> There are packages there for Squeeze, Wheezy and Sid... Jesse >>> packages are in the early stages >>> >>> >>> >>> On 5/13/13 4:18 PM, "Steven Ayre">> > wrote: >>> >>> Great news! Any idea when Wheezy packages will be added? >>> >>> -Steve >>> >>> >>> On 13 May 2013 21:52, Ken Rice>> > wrote: >>> >>> Hey Guys, >>> >>> Just a quick heads up, As you know last week we >>> released FreeSWITCH 1.2.9 and I said packages were >>> coming. >>> >>> We now have both YUM and Debian Repo's for you guys to >>> install from... >>> >>> For Debian check out >>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages >>> For Centos check out >>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation >>> >>> These will get you the stable branch for now... >>> >>> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the >>> deb's and rpm's for it >>> >>> >>> -- >>> Ken >>> _http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> _irc.freenode.net #freeswitch >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130513/d365d717/attachment-0001.html From alex at opensystems.net.au Tue May 14 06:57:07 2013 From: alex at opensystems.net.au (alex_ynema) Date: Mon, 13 May 2013 19:57:07 -0700 (PDT) Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: References: Message-ID: <1368500227717-7590648.post@n2.nabble.com> Would this work to limit the total concurrent outgoing calls for a system. I'm running newfies-dialer and we are pushing beyond our concurrent call limit of 100 concurrent calls. I was thinking somthing along these lines -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Limit-concurrent-calls-tp7589225p7590648.html Sent from the freeswitch-users mailing list archive at Nabble.com. From nbhatti at gmail.com Tue May 14 06:57:58 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 14 May 2013 05:57:58 +0300 Subject: [Freeswitch-users] Strange CDR entry In-Reply-To: <5191799A.1010000@freeswitch.org> References: <5191799A.1010000@freeswitch.org> Message-ID: <5191A836.8000402@gmail.com> We saw that hitting our servers too quite often. Looks like some kind of automated program. The script tries to call these numbers. It never got successful though. Thanks, -- Muhammad Naseer Bhatti I put the Who? in Mishehu wrote: > I bet that DID is hosted by Jawal. Jawal is a Palestinian telco, but > I think that 0598 is specifically cellular, so it's not limited to > Gaza. I've seen these hitting other servers on the Interwebs as well... > > -Yossi > > On 05/13/2013 09:35 AM, Ken Rice wrote: >> Ok look at the 3rd column... This is someone trying to figure out how >> to get calls thru your public context with various prefix codexs... >> Notices how all the calls end with 972598371070 <-- that?s the intl >> destination umber probably a phone in gaza (definitely in Isreali >> controlled areas, but I think I looked up this number or similar >> number in the past) >> >> To protect yourself, only allow calls to local extentsions from the >> public context... >> >> >> On 5/13/13 2:49 AM, "Steven Ayre" wrote: >> >> The call duration is 0 seconds so no they didn't get in. Any >> accepted call attempt will generate a CDR even if it was not >> answered. >> >> They're coming in on the 'public' context which'll mean an >> unauthenticated call on the external profile. That would be the >> context allowing you to receive calls from strangers. Providing >> that context doesn't allow them to dial out via providers that >> charge you this isn't an issue. >> >> -Steve >> >> >> >> >> >> >> On 13 May 2013 07:17, Bala Murugan Mahendran wrote: >> >> I have lot of entries like 1000,1001,1002 and so on, Is >> someone trying to get inside? I believe they didn't in yet >> but how come we have cdr log like this? >> >> "1004","1004","012972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c522fb02-bad5-11e2-8eaf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","013972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c5571b44-bad5-11e2-8eb3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","014972598371070","public","2013-05-12 >> 07:30:07","","2013-05-12 >> 07:30:07","0","0","NORMAL_CLEARING","c58eb428-bad5-11e2-8eb7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","010972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5c42d4c-bad5-11e2-8ebb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0061972598371070","public","2013-05-12 >> 07:30:08","","2013-05-12 >> 07:30:08","0","0","NORMAL_CLEARING","c5fe993c-bad5-11e2-8ebf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","0041972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c63f939c-bad5-11e2-8ec3-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","000972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c67a9f5a-bad5-11e2-8ec7-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","006972598371070","public","2013-05-12 >> 07:30:09","","2013-05-12 >> 07:30:09","0","0","NORMAL_CLEARING","c6afc2fc-bad5-11e2-8ecb-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","002972598371070","public","2013-05-12 >> 07:30:10","","2013-05-12 >> 07:30:10","0","0","NORMAL_CLEARING","c6ec0b5e-bad5-11e2-8ecf-7bfd76dfd5d7","","","G729","G729" >> "1004","1004","810972598371070","public","2013-05-12 >> 07:29:49","","2013-05-12 >> 07:29:49","0","0","NORMAL_CLEARING","ba7999a4-bad5-11e2-8b6f-7bfd76dfd5d7","","","G729","G729" >> >> >> Thanks, >> Bala >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> ------------------------------------------------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/83b44a60/attachment.html From krice at freeswitch.org Tue May 14 07:15:24 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 May 2013 22:15:24 -0500 Subject: [Freeswitch-users] Limit concurrent calls In-Reply-To: <1368500227717-7590648.post@n2.nabble.com> References: <1368500227717-7590648.post@n2.nabble.com> Message-ID: <99746631-38CF-4253-8199-D7122FFE5156@freeswitch.org> You know you can set max sessions on freeswitch each session is basically a call leg also check with newfies they probably have a method for that Sent from my iPhone On May 13, 2013, at 9:57 PM, alex_ynema wrote: > Would this work to limit the total concurrent outgoing calls for a system. > I'm running newfies-dialer and we are pushing beyond our concurrent call > limit of 100 concurrent calls. > > I was thinking somthing along these lines > > > > > > > > > > > data="{codec_string='PCMA'}sofia/gateway/${distributor(distributor1)}/$1" > loop="2" /> > > > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Limit-concurrent-calls-tp7589225p7590648.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://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 From alex at opensystems.net.au Tue May 14 07:31:58 2013 From: alex at opensystems.net.au (alex_ynema) Date: Mon, 13 May 2013 20:31:58 -0700 (PDT) Subject: [Freeswitch-users] =?utf-8?q?Error_message_=22_=5BERR=5D_switch?= =?utf-8?q?=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_events=E2=80=9C?= In-Reply-To: References: <1339645894596-7579817.post@n2.nabble.com> Message-ID: <1368502318906-7590651.post@n2.nabble.com> Where can I set this limit in Freeswitch. I've been hitting the limit quite regularly while using it to run call campaigns. Cheers -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ostolyar at netflix.com Tue May 14 03:42:47 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Mon, 13 May 2013 16:42:47 -0700 Subject: [Freeswitch-users] Dynamic external profile proxies In-Reply-To: <5191785F.50302@freeswitch.org> References: <5191785F.50302@freeswitch.org> Message-ID: Thanks Yossi, I looked into OpenSIPS - it's an option but it seems to also load the media server instances on load, which is something I think I can do with FreeSWITCH as well, so no reason to use two solutions. FreeSWITCH has its own SIP (Sofia) component which I was hoping to use to route SIP calls to other instances of FreeSWITCH. Am I missing something? Is there a way to dynamically create proxies for routing each call in OpenSIPS? Thank you *Oleg* On Mon, May 13, 2013 at 4:33 PM, I put the Who? in Mishehu < mishehu at freeswitch.org> wrote: > SIP is the signalling, and RTP is the media stream. For load-balancing > of SIP calls to different servers, you're probably better served by > OpenSIPS or a similar project (i.e. kamailio). FreeSWITCH can readily > interoperate with OpenSIPS and Kamailio. > > http://www.opensips.org > http://www.kamailio.org > > -Yossi > > > On 05/13/2013 10:40 AM, Oleg Stolyar wrote: > > I am very new to FreeSWITCH and would like to use it as a SIP load > balancer to other (media) FreeSWITCH instances. I was thinking of using > the external SIP profile and bridging incoming calls to the media instances > of FreeSWITCH using some rotation mechanism (round robin, least busy, etc) > > My question is this: Would I need to have a separate SIP profile for > each media FreeSWITCH server or can I define a profile in such a way that I > can dynamically specify the proxy for each call? > > Thank you > *Oleg* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130513/0d79c07f/attachment.html From john at telecube.com.au Tue May 14 05:21:09 2013 From: john at telecube.com.au (Telecube - John) Date: Tue, 14 May 2013 11:21:09 +1000 Subject: [Freeswitch-users] Recommended version for production Message-ID: <51919185.2090502@telecube.com.au> Hi, Can someone advise what would be the recommended version to use in a production environment please? -- John From jleung at v10networks.ca Tue May 14 08:15:35 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Mon, 13 May 2013 21:15:35 -0700 Subject: [Freeswitch-users] Recommended version for production In-Reply-To: <51919185.2090502@telecube.com.au> References: <51919185.2090502@telecube.com.au> Message-ID: <019901ce5059$af9a75d0$0ecf6170$@v10networks.ca> Use the v1.2.stable branch. That's the version recommended for production usage. > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of Telecube - John > Sent: Monday, May 13, 2013 6:21 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Recommended version for production > > Hi, > > Can someone advise what would be the recommended version to use in a > production environment please? > -- > John > > __________________________________________________________ > _______________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From bdfoster at davri.com Tue May 14 08:16:25 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 14 May 2013 00:16:25 -0400 Subject: [Freeswitch-users] Recommended version for production In-Reply-To: <51919185.2090502@telecube.com.au> References: <51919185.2090502@telecube.com.au> Message-ID: v1.2.stable or v1.2.9. http://wiki.freeswitch.org -BDF On May 14, 2013 12:08 AM, "Telecube - John" wrote: > Hi, > > Can someone advise what would be the recommended version to use in a > production environment please? > -- > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/4a354687/attachment.html From jaybinks at gmail.com Tue May 14 08:22:06 2013 From: jaybinks at gmail.com (jay binks) Date: Tue, 14 May 2013 14:22:06 +1000 Subject: [Freeswitch-users] Recommended version for production In-Reply-To: <51919185.2090502@telecube.com.au> References: <51919185.2090502@telecube.com.au> Message-ID: I use the 1.2 branch.. as for which sub version... you get to pick that one :) start with latest, and if you find problems go back. I think kens recent bump puts us at 1.2.9 ?? On 14 May 2013 11:21, Telecube - John wrote: > Hi, > > Can someone advise what would be the recommended version to use in a > production environment please? > -- > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/2c077ee0/attachment-0001.html From letterstack at gmail.com Tue May 14 08:59:32 2013 From: letterstack at gmail.com (Shiju V.Joseph) Date: Tue, 14 May 2013 10:29:32 +0530 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED Message-ID: eval and sofia_contact can find both the domain and registered user , then what could be wrong , it is driving me crazy ..is there anything I could have missed out Any hints would be highly helpful. Thanks shijujoe eval and sofia_contact returned the following freeswitch at internal> eval ${domain} ec2-54-242-200-55.compute-1.amazonaws.com freeswitch at internal> freeswitch at internal> sofia_contact */ 913288 at ec2-54-242-200-55.compute-1.amazonaws.com sofia/internal/sip:913288 at 107.20.242.28:38510 ;line=89c9013ed42887b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A107.20.242.28%3A38510%3E freeswitch at internal> On Wed, May 8, 2013 at 3:00 PM, Shiju V.Joseph wrote: > Hi All, > > I have been experimenting with an odbc based freeswitch cluster in amazon > ec2 , with opensips doing the load balancing function. > > I can make calls to mobile and landlines with out any issues with good > quality voice , but when i try to call extension to extension freeswith > shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot > in lists and wiki and fs jira , tried out different dial strings but with > out any results.Tried this one > "dial-string" value="{presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & > this ${sofia_contact(*/${dialed_user}@${dialed_domain})} > > I have copied the siptrace at http://pastebin.com/Qv79tjXK > > Appreciate any help in this regard > > Thanks > -- > shijujoe > -- Shiju V.Joseph -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/a517bd02/attachment.html From ashish at nms.co.in Tue May 14 09:31:54 2013 From: ashish at nms.co.in (Ashish gautam) Date: Tue, 14 May 2013 11:01:54 +0530 Subject: [Freeswitch-users] hangup extension In-Reply-To: <5190AC8A.1080903@telecube.com.au> References: <5190AC8A.1080903@telecube.com.au> Message-ID: Session gets dissolved in hangupHook and we cannot do anything on the basis of UUID On Mon, May 13, 2013 at 2:34 PM, Telecube - John wrote: > If you are using Lua script there is ' session:setHangupHook' detailed > here: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook > > I assume there's something similar in the xml configs > > Regards, > John > > > > On 13/05/13 5:20 PM, Ashish gautam wrote: > > Hi, > > Is there anything like hangup extension which gets executed after the > call is hung up for which the UUID is same and based on that I can do > things for the same call like we have in Asterisk? > > Thanks. > > Regards, > > --Ashish > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130514/538b972e/attachment.html From dujinfang at gmail.com Tue May 14 09:41:01 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 14 May 2013 13:41:01 +0800 Subject: [Freeswitch-users] eavesdrop not sending video In-Reply-To: References: Message-ID: The wiki page is right, it also said you should use the video-media-bug branch. Since there's a BIG refactor on media in the master version, we need some time to make this up to date, also maybe we can merge into master if there's enough interest. On Tuesday, May 14, 2013 at 4:47 AM, Ken Rice wrote: > Re: [Freeswitch-users] eavesdrop not sending video Actually I believe the wiki is wrong here, and I confirmed with Anthony, eavesdrop does not work on video... > > The is probably in mod_commands but I am not sure > > > On 5/13/13 2:31 PM, "D D" wrote: > > > The wiki page http://wiki.freeswitch.org/wiki/Video-recording says that video eavesdrop should work. > > > > Which source file should I check in order to verify the implementation? > > > > > > > > > > > > > > > > From: Ken Rice > > To: FreeSWITCH Users Help > > Sent: Monday, May 13, 2013 11:19 AM > > Subject: Re: [Freeswitch-users] eavesdrop not sending video > > > > > > > > Re: [Freeswitch-users] eavesdrop not sending video > > Eavesdrop probably does not support sending the video stream as well... If I recall correctly, it only taps the audio stream... Possible solutions, open a jira and offer a bounty for someone to look at adding support for video > > > > On 5/13/13 10:54 AM, "D D" wrote: > > > > > Hi, > > > > > > I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. > > > > > > I have followed the directions described at http://wiki.freeswitch.org/wiki/Video-recording > > > (including a rebuild after doing the git checkout of video-media-bug), and I can successfully > > > use record_fsv to record the video to a file. > > > > > > Any ideas why the eavesdrop does not send the video? > > > > > > Thanks, > > > David > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.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 > > -- > Ken > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > irc.freenode.net (http://irc.freenode.net) #freeswitch > _________________________________________________________________________ > 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://wiki.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/20130514/6e4a19f1/attachment.html From john at telecube.com.au Tue May 14 08:52:11 2013 From: john at telecube.com.au (Telecube - John) Date: Tue, 14 May 2013 14:52:11 +1000 Subject: [Freeswitch-users] Recommended version for production In-Reply-To: References: <51919185.2090502@telecube.com.au> Message-ID: <5191C2FB.9000804@telecube.com.au> Hi Jay, Thanks, I'll take a look at the yum repo and do some tests on an install of 1.2.9 from there. Cheers, John On 14/05/13 2:22 PM, jay binks wrote: > I use the 1.2 branch.. > as for which sub version... you get to pick that one :) > > start with latest, and if you find problems go back. > I think kens recent bump puts us at 1.2.9 ?? > > > On 14 May 2013 11:21, Telecube - John > wrote: > > Hi, > > Can someone advise what would be the recommended version to use in a > production environment please? > -- > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Sincerely > > Jay > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/19e14cdb/attachment-0001.html From alex at digitalmail.com Tue May 14 12:16:20 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 14 May 2013 09:16:20 +0100 Subject: [Freeswitch-users] Changes between versions In-Reply-To: <51914144.5070901@freeswitch.org> References: <51914144.5070901@freeswitch.org> Message-ID: <5191F2D4.8040202@digitalmail.com> ...and I'd run that locally in the /usr/src/freeswitch directory? > On 13-05-13 01:58 PM, Ken Rice wrote: >> Git log tag..tag >> I think >> > i think it's > git diff tag..tag > > -Ray > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2241 / Virus Database: 3162/5820 - Release Date: 05/13/13 > > From alex at digitalmail.com Tue May 14 16:07:53 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 14 May 2013 13:07:53 +0100 Subject: [Freeswitch-users] Idiot question concerning internal calls Message-ID: <51922919.8000407@digitalmail.com> I've just discovered a yawning hole in my knowledge of Freeswitch... When I make an internal call, I would like to be able to set variables that are originator specific. I had previously done this in the directory part of conf with things like this: ... But when I try to extract those variables in a lua script (eg. the extension handling "201"), they don't seem to be there. It's as though they weren't exported from A-Leg to B-Leg. What am I missing? How can I arrange for variables set in the to be readable in the b-leg dialplan? From avi at avimarcus.net Tue May 14 16:37:41 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 14 May 2013 15:37:41 +0300 Subject: [Freeswitch-users] Idiot question concerning internal calls In-Reply-To: <51922919.8000407@digitalmail.com> References: <51922919.8000407@digitalmail.com> Message-ID: How are you creating a B leg? Bridging to user/ ? -Avi On Tue, May 14, 2013 at 3:07 PM, Alex Lake wrote: > I've just discovered a yawning hole in my knowledge of Freeswitch... > > When I make an internal call, I would like to be able to set variables > that are originator specific. > > I had previously done this in the directory part of conf with things > like this: > > > > ... > > > > > > > > > > > But when I try to extract those variables in a lua script (eg. the > extension handling "201"), they don't seem to be there. > > It's as though they weren't exported from A-Leg to B-Leg. > > What am I missing? > > How can I arrange for variables set in the to be readable in the > b-leg dialplan? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/934bd4ff/attachment.html From raimund.sacherer at logitravel.com Tue May 14 16:58:36 2013 From: raimund.sacherer at logitravel.com (Raimund Sacherer) Date: Tue, 14 May 2013 14:58:36 +0200 (CEST) Subject: [Freeswitch-users] Idiot question concerning internal calls In-Reply-To: <16495550.4842.1368536314023.JavaMail.javamailuser@localhost> Message-ID: <25503894.4843.1368536315159.JavaMail.javamailuser@localhost> without having more information on the rest of your dialplan, i just will shout out: export, check the export variables or dialplan app, i use it in various scenarios when I have to make sure variables are available in all cdrs. ----- Original Message ----- From: "Alex Lake" To: "FreeSWITCH Users Help" Sent: Martes, 14 de Mayo 2013 14:07:53 Subject: [Freeswitch-users] Idiot question concerning internal calls I've just discovered a yawning hole in my knowledge of Freeswitch... When I make an internal call, I would like to be able to set variables that are originator specific. I had previously done this in the directory part of conf with things like this: ... But when I try to extract those variables in a lua script (eg. the extension handling "201"), they don't seem to be there. It's as though they weren't exported from A-Leg to B-Leg. What am I missing? How can I arrange for variables set in the to be readable in the b-leg dialplan? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Logitravel.com Raimund Sacherer Sistemas Agencia de Viajes Online www.logitravel.com Edificio Logitravel, Parcela 3B (Parc Bit) Ctra. Palma - Valldemossa km 7,4 | 07121 Palma de Mallorca Tel 902 366 847 | Fax 971 213 495 S?guenos en: Facebook de Logitravel Twitter de Logitravel Blog de Logitravel Logitravel en Youtube Logitravel en Foursquare Descarga nuestras aplicaciones para m?vil Logitravel.com Este correo electr?nico y, en su caso, cualquier fichero anexo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de LOGITRAVEL S.L.. En caso de haber recibido este correo electr?nico por error, se ruega notif?quese inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. Al mismo tiempo LA EMPRESA le recuerda que sus datos forman o formar?n parte de un fichero registrado como CLIENTES con n?mero de inscripci?n 2070610043 en la Agencia General de Protecci?n de Datos, propiedad de la empresa LOGITRAVEL, con domicilio en Edificio Logitravel, Ctra. Palma - Valldemosa km 7,4, Parc Bit, Palma de Mallorca. Usted tiene derecho de acceso, oposici?n, rectificaci?n y cancelaci?n a estos datos que deber? ejercer mediante escrito a la direcci?n anteriormente citada. From andrew at cassidywebservices.co.uk Tue May 14 17:23:24 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 14 May 2013 14:23:24 +0100 Subject: [Freeswitch-users] Faxing and loopback fun! Message-ID: Hi all, I was just randomly thinking about something I was playing with a while ago and issues surrounding it. Basically, I had a web page that you uploaded a PDF too, which was then faxed out by FreeSWITCH using ESL using a loopback dialstring as it isn't necessarily easy to determine the routes that should be used to send the fax. Obviously this had some issues, didn't like T.38, etc. My current workaround is to store all the same routes in a database and loop through them all as part of the upload page to determine the sofia dialstring I should use as part of the originate command. Is there an easier way I could do this directly in FreeSWITCH that I've completely missed? -- *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/20130514/97349bf7/attachment.html From alex at digitalmail.com Tue May 14 18:28:40 2013 From: alex at digitalmail.com (Alex Lake) Date: Tue, 14 May 2013 15:28:40 +0100 Subject: [Freeswitch-users] Idiot question concerning internal calls In-Reply-To: References: <51922919.8000407@digitalmail.com> Message-ID: <51924A18.1050107@digitalmail.com> Aaargghh! Sorry guys - I had a little lua function that "de-nilled" my variables and it had a typo in it that meant it always returned an empty string.... Doh... > How are you creating a B leg? Bridging to user/ ? > -Avi > > On Tue, May 14, 2013 at 3:07 PM, Alex Lake > wrote: > > I've just discovered a yawning hole in my knowledge of Freeswitch... > > When I make an internal call, I would like to be able to set variables > that are originator specific. > > I had previously done this in the directory part of conf with things > like this: > > > > ... > > > > > > > > > > > But when I try to extract those variables in a lua script (eg. the > extension handling "201"), they don't seem to be there. > > It's as though they weren't exported from A-Leg to B-Leg. > > What am I missing? > > How can I arrange for variables set in the to be readable > in the > b-leg dialplan? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2241 / Virus Database: 3162/5822 - Release Date: 05/13/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/1e76dcf5/attachment-0001.html From bdfoster at davri.com Tue May 14 19:39:07 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 14 May 2013 11:39:07 -0400 Subject: [Freeswitch-users] Faxing and loopback fun! In-Reply-To: References: Message-ID: Use mod_lcr. Works good for that kind of thing. On May 14, 2013 9:29 AM, "Andrew Cassidy" wrote: > Hi all, > > I was just randomly thinking about something I was playing with a while > ago and issues surrounding it. Basically, I had a web page that you > uploaded a PDF too, which was then faxed out by FreeSWITCH using ESL using > a loopback dialstring as it isn't necessarily easy to determine the routes > that should be used to send the fax. Obviously this had some issues, > didn't like T.38, etc. > > My current workaround is to store all the same routes in a database and > loop through them all as part of the upload page to determine the sofia > dialstring I should use as part of the originate command. Is there an > easier way I could do this directly in FreeSWITCH that I've completely > missed? > > -- > *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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/c37e459a/attachment.html From anthony.minessale at gmail.com Tue May 14 19:43:10 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 May 2013 10:43:10 -0500 Subject: [Freeswitch-users] =?windows-1252?q?Error_message_=22_=5BERR=5D_s?= =?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?= =?windows-1252?q?ts=93?= In-Reply-To: <1368502318906-7590651.post@n2.nabble.com> References: <1339645894596-7579817.post@n2.nabble.com> <1368502318906-7590651.post@n2.nabble.com> Message-ID: Its the 3rd arg to the EventConsumer constructor and it defaults to 5000 On Mon, May 13, 2013 at 10:31 PM, alex_ynema wrote: > Where can I set this limit in Freeswitch. > I've been hitting the limit quite regularly while using it to run call > campaigns. > > Cheers > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.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://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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/b6ed0bbc/attachment.html From dgarcia at anew.com.ve Tue May 14 19:52:23 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 14 May 2013 11:22:23 -0430 Subject: [Freeswitch-users] Advice using mod_curl In-Reply-To: References: <518D413D.301@anew.com.ve> Message-ID: <51925DB7.3040507@anew.com.ve> Thanks Cal for your answer, Well, The web service is not in my control. The slowly response could be due to web server issues, network, internet conection, webservice itself, etc. In my country, quck and stable internet access is not granted as is in other countries. I suspect is due to network and internet conections. Before open this thread, I got some ideas in how to do it. I just wanted to know how other fs user do it. I tried to put the call in hold: 1. Tring to transfer the call to music_hod (9664) ... but I got stuck, how I retrieve/recover/unhold the call? 2. I tried using other fs way like uuid_hold, the call seems to be somewhere, no music is played, and then I got stuck, how I retrieve/recover/unhold the call? How uuid_hold should be used? I will try httpapi. Any suggestion, will be welcome On 5/10/2013 2:43 PM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello, > > Can I ask why your web service is taking several seconds to respond? > Could you not speed this up? > > As far as I'm aware, telling mod_curl to play music when a request is > being executed is really not the right way to look at it. If you need > to retrieve data from an external service that will take several > seconds, you would place the call into hold or into a parking queue, > execute the request, then come back.. however you'd do this within the > mod_curl response, rather than it happening on a mod_curl call itself. > For example, you could get the mod_curl to response immediately, then > use a mixture of httpapi/includes/parking to achieve your goal.. > > Perhaps someone else might be able to offer a cleaner solution, but > that's the best I can think of at the moment. > > Hope this helps! > > Cal > > On Fri, May 10, 2013 at 7:49 PM, Saugort Dario Garcia Tovar > > wrote: > > Hi, > > I have lua script. the script is similar to the one in > http://wiki.freeswitch.org/wiki/Mod_curl (lua usage) > > The script access a web service and get some data. The script work > fine for my needs. However, sometimes when the script access the > web service take several seconds to get the data, and the caller > ear silence. Is it possible to play a sound file as long the > script try to get the data from the web service? > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2241 / Virus Database: 3162/5814 - Release Date: 05/10/13 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/3eaa6be0/attachment-0001.html From anthony.minessale at gmail.com Tue May 14 19:52:30 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 14 May 2013 10:52:30 -0500 Subject: [Freeswitch-users] TCP ACK Ping In-Reply-To: <51914101.50504@5ninesolutions.com> References: <51900FA2.9020509@5ninesolutions.com> <51914101.50504@5ninesolutions.com> Message-ID: We've run into that too, something goofy in 4.x firmware. The sofia profile param tcp-keepalive can be set to like 60000 to send a packet once a minute to keep them open. I think I also added the flag to the socket in linux to do this tcp ping pong stuff but it appears that the polycom is doing it some other way and even with a steady flow of these, it gives up and disconnects. So this param above may fix it but I think the polycom is actually not behaving properly and we have to pay for it now with scheduler overhead in the lib. On Mon, May 13, 2013 at 2:37 PM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > Understood, I though it was quite strange to see these unsolicited ACKs > from the Polycoms followed by a RST. > > Thanks, > Spencer > > > > On 05/13/2013 02:07 PM, Steven Schoch wrote: > > On Sun, May 12, 2013 at 2:54 PM, Spencer Thomason < > spencer at 5ninesolutions.com> wrote: > >> The phones periodically send a TCP ACK ping to >> Freeswitch which goes unanswered and then phone then tears down the TCP >> connection. I'm not at all a TCP expert but should Freeswitch be >> responding to this unsolicited ACK? > > > I've been working with TCP for 25 years, and this is the first I have > heard of ACK ping. A quick Google search told me that this technique > involves sending a normal TCP ACK packet (which all TCP packets except the > initial one have, by the way) to a random TCP port. The host will then > respond with a RST packet. (Which means reset the connection because the > ACK was not sent to an established connection.) > > However, a firewall may filter and discard these random ACK packets. > Since they're not part of "normal" TCP, no one cares. > > In answer to your question, this is in the TCP layer much deeper than > Freeswitch. There is nothing Freeswitch can do at this level. > > -- > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/329d7c2b/attachment.html From cesar.bermudez at gmail.com Tue May 14 20:38:59 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 10:38:59 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX Message-ID: Hi all. What its the best GUI for use FS in pbx? and why? bluebox? fusion? any other? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/233aaff8/attachment.html From jpyle at fidelityvoice.com Tue May 14 21:27:36 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Tue, 14 May 2013 13:27:36 -0400 Subject: [Freeswitch-users] pass "called" RPID from B-leg to A-leg In-Reply-To: <5191534B.9060706@fidelityvoice.com> References: <5191534B.9060706@fidelityvoice.com> Message-ID: <51927408.4070804@fidelityvoice.com> I think I'm making some progress. The A-leg "effective_callee_id_name" appears on the caller's handset during provisional response / ringing time. The B-leg "effective_callee_id_name" appears on the caller's handset when the call is connected. Knowing this, is it possible capture the callee name/number from the RPID in the received 18x message to populate the right variable at the right time? - Jeff Jeff Pyle wrote: > Hello, > > Searching the archives I've seen this question asked a few times...but > never answered. Or, at least not in a way that my feeble neurons can > process. > > A-leg is bridged to B-leg with: > data="sofia/gateway/gw_name/${destination_number}"/> > > The B-leg responds with a Remote-Party-Id header in a 180 Ringing. > How can I pass this information into the 180 Ringing that gets send > back to the A-leg? > > It seems I may need to set sip_callee_id_number/name at some point, > but I'm not sure how to get at the B-leg data to set those variables. > > > - Jeff > From luis.daniel.lucio at gmail.com Tue May 14 22:02:28 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 14 May 2013 14:02:28 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: fusion, when configured correctly you will love it 2013/5/14 Cesar Bermudez > Hi all. > > What its the best GUI for use FS in pbx? and why? > bluebox? fusion? any other? > > Regards. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/5bfb7cc7/attachment.html From cesar.bermudez at gmail.com Tue May 14 22:07:14 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 12:07:14 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Any hints/tips/tricks to configure correctly? Regards. On Tue, May 14, 2013 at 12:02 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > fusion, when configured correctly you will love it > > > 2013/5/14 Cesar Bermudez > >> Hi all. >> >> What its the best GUI for use FS in pbx? and why? >> bluebox? fusion? any other? >> >> Regards. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/863dd240/attachment-0001.html From j at 2600hz.com Tue May 14 22:07:59 2013 From: j at 2600hz.com (Joshua Goldbard) Date: Tue, 14 May 2013 18:07:59 +0000 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: We're looking to do some work on BlueBox in the future. As it stands right now it's a very good GUI but we think it could deal with an update or two. I've heard good things about Fusion, but my boss is the BlueBox author so I feel I have to speak up when people talk about FreeSWITCH GUIs :D. Cheers, Joshua Joshua Goldbard VP of Marketing, 2600hz 116 Natoma Street, Floor 2 San Francisco, CA, 94104 415.886.7923 | j at 2600hz.com On May 14, 2013, at 11:02 AM, Luis Daniel Lucio Quiroz > wrote: fusion, when configured correctly you will love it 2013/5/14 Cesar Bermudez > Hi all. What its the best GUI for use FS in pbx? and why? bluebox? fusion? any other? Regards. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130514/c7a50dc7/attachment.html From vipkilla at gmail.com Tue May 14 22:14:48 2013 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 14 May 2013 14:14:48 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: don't rely on a gui. learn how to configure FS manually, then build your own GUI :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/b48ef903/attachment.html From andrew at cassidywebservices.co.uk Tue May 14 22:17:26 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 14 May 2013 19:17:26 +0100 Subject: [Freeswitch-users] Faxing and loopback fun! In-Reply-To: References: Message-ID: Yeah it's slightly less flexible than what I had in mind with the original design. However I bet it's loads better performance wise and perhaps a logical trade-off. On 14 May 2013 16:39, Brian Foster wrote: > Use mod_lcr. Works good for that kind of thing. > On May 14, 2013 9:29 AM, "Andrew Cassidy" > wrote: > >> Hi all, >> >> I was just randomly thinking about something I was playing with a while >> ago and issues surrounding it. Basically, I had a web page that you >> uploaded a PDF too, which was then faxed out by FreeSWITCH using ESL using >> a loopback dialstring as it isn't necessarily easy to determine the routes >> that should be used to send the fax. Obviously this had some issues, >> didn't like T.38, etc. >> >> My current workaround is to store all the same routes in a database and >> loop through them all as part of the upload page to determine the sofia >> dialstring I should use as part of the originate command. Is there an >> easier way I could do this directly in FreeSWITCH that I've completely >> missed? >> >> -- >> *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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *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/20130514/6e9a08c7/attachment.html From cesar.bermudez at gmail.com Tue May 14 22:20:55 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 12:20:55 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: I see the bluebox, but is live that project? i visited the git and the code is 2 year old? for that reason i thinked "this is death" ... what is the status of the bluebox? On Tue, May 14, 2013 at 12:07 PM, Joshua Goldbard wrote: > We're looking to do some work on BlueBox in the future. As it stands > right now it's a very good GUI but we think it could deal with an update or > two. > > I've heard good things about Fusion, but my boss is the BlueBox author > so I feel I have to speak up when people talk about FreeSWITCH GUIs :D. > > Cheers, > Joshua > > Joshua Goldbard > VP of Marketing, 2600hz > > 116 Natoma Street, Floor 2 > San Francisco, CA, 94104 > 415.886.7923 | j at 2600hz.com > > On May 14, 2013, at 11:02 AM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> > wrote: > > fusion, when configured correctly you will love it > > > 2013/5/14 Cesar Bermudez > >> Hi all. >> >> What its the best GUI for use FS in pbx? and why? >> bluebox? fusion? any other? >> >> Regards. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/6b756ea1/attachment-0001.html From cesar.bermudez at gmail.com Tue May 14 22:22:47 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 12:22:47 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: thats the idea, but for now we will start using one of these two ( if there are no more options ) then maybe 2 ways, start using the programmers to build one ( i think this will take more time ), or just contribute back to one of the actual working gui. regards. On Tue, May 14, 2013 at 12:14 PM, Vik Killa wrote: > don't rely on a gui. > learn how to configure FS manually, then build your own GUI :) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/f4ebdb24/attachment.html From bpriddy at bryantschools.org Tue May 14 22:41:55 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Tue, 14 May 2013 13:41:55 -0500 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: We use FusionPBX in production. Bluebox is too out of date. :( wiki.fusionpbx.com Mark J Crane is the Author He is on #fusionpbx on freenode as mcrane On Tue, May 14, 2013 at 1:22 PM, Cesar Bermudez wrote: > thats the idea, but for now we will start using one of these two ( if > there are no more options ) > then maybe 2 ways, start using the programmers to build one ( i think this > will take more time ), > or just contribute back to one of the actual working gui. > > regards. > > On Tue, May 14, 2013 at 12:14 PM, Vik Killa wrote: > >> don't rely on a gui. >> learn how to configure FS manually, then build your own GUI :) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/7c3a034b/attachment.html From cesar.bermudez at gmail.com Tue May 14 22:46:42 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 12:46:42 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Thx for your answer Blake. On Tue, May 14, 2013 at 12:41 PM, Blake Priddy wrote: > We use FusionPBX in production. Bluebox is too out of date. :( > > wiki.fusionpbx.com > > Mark J Crane is the Author > > He is on #fusionpbx on freenode as mcrane > > > On Tue, May 14, 2013 at 1:22 PM, Cesar Bermudez wrote: > >> thats the idea, but for now we will start using one of these two ( if >> there are no more options ) >> then maybe 2 ways, start using the programmers to build one ( i think >> this will take more time ), >> or just contribute back to one of the actual working gui. >> >> regards. >> >> On Tue, May 14, 2013 at 12:14 PM, Vik Killa wrote: >> >>> don't rely on a gui. >>> learn how to configure FS manually, then build your own GUI :) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/ac72ee3a/attachment.html From cesar.bermudez at gmail.com Tue May 14 22:50:00 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 14 May 2013 12:50:00 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: For that reason i asked, i dont installed anything yet, but looking at the websites, and gits, i have the same feeling, bluebox outdated, and fusionpbx still active developement, just sharing my feeling about this, and asking in general. Regards. On Tue, May 14, 2013 at 12:41 PM, Blake Priddy wrote: > We use FusionPBX in production. Bluebox is too out of date. :( > > wiki.fusionpbx.com > > Mark J Crane is the Author > > He is on #fusionpbx on freenode as mcrane > > > On Tue, May 14, 2013 at 1:22 PM, Cesar Bermudez wrote: > >> thats the idea, but for now we will start using one of these two ( if >> there are no more options ) >> then maybe 2 ways, start using the programmers to build one ( i think >> this will take more time ), >> or just contribute back to one of the actual working gui. >> >> regards. >> >> On Tue, May 14, 2013 at 12:14 PM, Vik Killa wrote: >> >>> don't rely on a gui. >>> learn how to configure FS manually, then build your own GUI :) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/3e1bfc73/attachment-0001.html From sdevoy at bizfocused.com Wed May 15 02:08:23 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 14 May 2013 18:08:23 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! Message-ID: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> HI all, My mother has made a request and Mom's always get our best effort right? For those of you outside the US, we have a national Do Not Call Registry: www.donotcall.gov. You register your number there and in 30 days or less you are on the list. Telemarketers are required by law to check the list before calling and not call if you are on it. It does certainly help, but overseas call centers tend to ignore it and of course the excluded groups from the law "charities, political organizations, and telephone surveyors" may still call you. So it is simple enough for her to tell me the number that called and I add it to my dial plan that plays a recording that politely says "go pound sand." So help me take this to the next level: 1. What should I be returning to the caller's switch (like temp unavail) instead of a recording? How do I do that? 2. She has an ATA and analog phone (so extra features are limited), but I would love to be able to give her some kind of option like "hang up and dial *666 to add them to the list of band numbers" or maybe "hook-flash dial *666 and hang up" to xfer them to the dial plan that does that? All ideas are welcome and certainly appreciated. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/23c7181e/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/23c7181e/attachment.gif From sirimmfs at gmail.com Wed May 15 02:08:23 2013 From: sirimmfs at gmail.com (Siri MM) Date: Wed, 15 May 2013 08:08:23 +1000 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: References: Message-ID: Hey Guys, Would appreciate any inputs! Thanks. On Tue, May 14, 2013 at 10:31 AM, Siri MM wrote: > Hello All, > > I am facing an issue with freeswitch installed on my device. For some > reason, IP address on my device sometimes seems to be 'fluctuating' between > the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this > change everytime, and reconfigure accordingly. However, during one such > occassion, it hasn't brought up the internal profile, and isn't binding to > port 5060. No other program uses this port on my device. > > Working log: > http://pastebin.freeswitch.org/20912 > > Not-working log: > http://pastebin.freeswitch.org/20915 > > Would appreciate any inputs > > Thanks! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/9c728b11/attachment.html From msc at freeswitch.org Wed May 15 02:10:53 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 15:10:53 -0700 Subject: [Freeswitch-users] =?windows-1252?q?Error_message_=22_=5BERR=5D_s?= =?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?= =?windows-1252?q?ts=93?= In-Reply-To: <1368502318906-7590651.post@n2.nabble.com> References: <1339645894596-7579817.post@n2.nabble.com> <1368502318906-7590651.post@n2.nabble.com> Message-ID: Doesn't this also mean that your program can't keep up with all of the events coming in? You may need to make your program more efficient or slow down the speed of the campaign. Have you been able to determine why your program isn't consuming events quickly enough? -MC On Mon, May 13, 2013 at 8:31 PM, alex_ynema wrote: > Where can I set this limit in Freeswitch. > I've been hitting the limit quite regularly while using it to run call > campaigns. > > Cheers > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.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://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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/367e1626/attachment.html From steveayre at gmail.com Wed May 15 02:11:48 2013 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 14 May 2013 23:11:48 +0100 Subject: [Freeswitch-users] =?windows-1252?q?Error_message_=22_=5BERR=5D_s?= =?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?= =?windows-1252?q?ts=93?= In-Reply-To: References: <1339645894596-7579817.post@n2.nabble.com> <1368502318906-7590651.post@n2.nabble.com> Message-ID: Don't set it so high that you run out of memory. If your queue is growing now, with a higher limit it's going to grow more so. I suggest you look at how to speed up your consuming. Some ideas: - Use a new MySQL version - 5.5/5.6 (there will be numerous performance improvements over older versions) - Use the innodb engine (row locking this will better performance as multiple connections can use the table at the same time) - Use multiple DB connections at the same time (which will allow multiple DB threads therefore using all the cores of your CPU) - Use a large innodb buffer size (ideally this should be as large as your data set or at least what you actively use - this'll mean you don't need to read from disk as often and so reads *and* writes will be faster) - Use EXPLAIN to tune your queries and indexes. Don't underindex (so you can find/update data quickly) but also don't overindex (which will waste cpu and disk updating indexes that are rarely/never used). There're numerous other ways to tune it too. -Steve On 14 May 2013 16:43, Anthony Minessale wrote: > Its the 3rd arg to the EventConsumer constructor and it defaults to 5000 > > > > On Mon, May 13, 2013 at 10:31 PM, alex_ynema wrote: > >> Where can I set this limit in Freeswitch. >> I've been hitting the limit quite regularly while using it to run call >> campaigns. >> >> Cheers >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.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://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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/93324a65/attachment-0001.html From msc at freeswitch.org Wed May 15 02:12:49 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 15:12:49 -0700 Subject: [Freeswitch-users] hangup extension In-Reply-To: References: <5190AC8A.1080903@telecube.com.au> Message-ID: On Mon, May 13, 2013 at 10:31 PM, Ashish gautam wrote: > Session gets dissolved in hangupHook and we cannot do anything on the > basis of UUID Normally yes, but that's why we have the chan var session_in_hangup_hook. Try setting that to true and then using your api_hangup_hook. -MC > > On Mon, May 13, 2013 at 2:34 PM, Telecube - John wrote: > >> If you are using Lua script there is ' session:setHangupHook' detailed >> here: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook >> >> I assume there's something similar in the xml configs >> >> Regards, >> John >> >> >> >> On 13/05/13 5:20 PM, Ashish gautam wrote: >> >> Hi, >> >> Is there anything like hangup extension which gets executed after the >> call is hung up for which the UUID is same and based on that I can do >> things for the same call like we have in Asterisk? >> >> Thanks. >> >> Regards, >> >> --Ashish >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/e9dee96b/attachment.html From msc at freeswitch.org Wed May 15 03:04:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 16:04:45 -0700 Subject: [Freeswitch-users] eavesdrop not sending video In-Reply-To: References: <1368473483.38110.YahooMailNeo@web120701.mail.ne1.yahoo.com> Message-ID: If it's in the dialplan then try looking in src/mod/appilcations/mod_dptools/mod_dptools.c. -MC On Mon, May 13, 2013 at 1:47 PM, Ken Rice wrote: > Actually I believe the wiki is wrong here, and I confirmed with Anthony, > eavesdrop does not work on video... > > The is probably in mod_commands but I am not sure > > > > On 5/13/13 2:31 PM, "D D" wrote: > > The wiki page http://wiki.freeswitch.org/wiki/Video-recording says that > video eavesdrop should work. > > Which source file should I check in order to verify the implementation? > > > > > > > > ------------------------------ > *From:* Ken Rice > *To:* FreeSWITCH Users Help > *Sent:* Monday, May 13, 2013 11:19 AM > *Subject:* Re: [Freeswitch-users] eavesdrop not sending video > > > > Re: [Freeswitch-users] eavesdrop not sending video > Eavesdrop probably does not support sending the video stream as well... If > I recall correctly, it only taps the audio stream... Possible solutions, > open a jira and offer a bounty for someone to look at adding support for > video > > On 5/13/13 10:54 AM, "D D" wrote: > > Hi, > > I tried to use eavesdrop to capture the video of a parked call, but I only > receive the audio. > > I have followed the directions described at > http://wiki.freeswitch.org/wiki/Video-recording > (including a rebuild after doing the git checkout of video-media-bug), and > I can successfully > use record_fsv to record the video to a file. > > Any ideas why the eavesdrop does not send the video? > > Thanks, > David > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/cb78cfea/attachment.html From msc at freeswitch.org Wed May 15 03:27:21 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 16:27:21 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: A belated happy new week to you all! As Ken mentioned in his email yesterday, the big news from last week is, of course, the release of FreeSWITCH 1.2.9 stable. This is now the "latest stable" version of FreeSWITCH and the one most recommended for production use. The tarball is available from theusual location . The CentOS and Debian packages have also been updated to use the latest stable. On last week's conference call we welcomed Dan from the CGRateS project . CGRateS is a relatively new project that handles call rating for carrier-grade operations. We are happy to have CGRateS (an open-source project itself) as part of the FreeSWITCH ecosystem. On this week's call we look at another VoIP-related OSS project: VoIPMonitor.org. This software performs a number of monitoring functions for VoIP implementations. Please join us tomorrow at 1PM EDT, 10AM PDT for an interesting discussion. One ClueCon reminder: We are still accepting speaking proposals, however time is running out so please send them in as soon as possible. We hope to have the ClueCon 2013 schedule released in the next few weeks. Take care and have a great week! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/561a1be3/attachment-0001.html From msc at freeswitch.org Wed May 15 03:46:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 16:46:20 -0700 Subject: [Freeswitch-users] Originate Failed. Cause: USER_NOT_REGISTERED In-Reply-To: References: Message-ID: Shiju, Who is the called party here? Is it 506393? If so that's the user you need to look up. Judging by the trace you sent I'd assume that 913288 is the calling party. It doesn't matter if the calling party is registered - it matters if the called party is registered. Do your sofia_contact tests on 506393 and see what happens. -MC On Mon, May 13, 2013 at 9:59 PM, Shiju V.Joseph wrote: > eval and sofia_contact can find both the domain and registered user , then > what could be wrong , it is driving me crazy ..is there anything I could > have missed out > > Any hints would be highly helpful. > > Thanks > shijujoe > > eval and sofia_contact returned the following > > freeswitch at internal> eval ${domain} > ec2-54-242-200-55.compute-1.amazonaws.com > freeswitch at internal> > > freeswitch at internal> sofia_contact */ > 913288 at ec2-54-242-200-55.compute-1.amazonaws.com > sofia/internal/sip:913288 at 107.20.242.28:38510 > ;line=89c9013ed42887b;fs_path=%3Csip%3A107.21.222.6%3A5060%3Blr%3Breceived%3Dsip%3A107.20.242.28%3A38510%3E > freeswitch at internal> > > > > On Wed, May 8, 2013 at 3:00 PM, Shiju V.Joseph wrote: > >> Hi All, >> >> I have been experimenting with an odbc based freeswitch cluster in amazon >> ec2 , with opensips doing the load balancing function. >> >> I can make calls to mobile and landlines with out any issues with good >> quality voice , but when i try to call extension to extension freeswith >> shows Originate Failed. Cause: USER_NOT_REGISTERED , i searched a lot >> in lists and wiki and fs jira , tried out different dial strings but with >> out any results.Tried this one >> "dial-string" value="{presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" & >> this ${sofia_contact(*/${dialed_user}@${dialed_domain})} >> >> I have copied the siptrace at http://pastebin.com/Qv79tjXK >> >> Appreciate any help in this regard >> >> Thanks >> -- >> shijujoe >> > > > > -- > Shiju V.Joseph > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/96c04280/attachment.html From msc at freeswitch.org Wed May 15 03:53:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 16:53:22 -0700 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> Message-ID: In the example config there's already a "call return" feature that utilizes the last received caller ID value. If there was a valid caller ID number sent then it will be stored in the local database, which is where the call_return extension gets it from. -MC P.S. - If you haven't already started using the FS local database then check it out. Try the "hash_dump all" command at fs_cli to get an idea of what gets stored. On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: > HI all,**** > > ** ** > > My mother has made a request and Mom?s always get our best effort right?** > ** > > ** ** > > For those of you outside the US, we have a national Do Not Call Registry: > www.donotcall.gov. You register your number there and in 30 days or less > you are on the list. Telemarketers are required by law to check the list > before calling and not call if you are on it.**** > > ** ** > > It does certainly help, but overseas call centers tend to ignore it and of > course the excluded groups from the law ?*charities, political > organizations, and telephone surveyors*? may still call you.**** > > ** ** > > So it is simple enough for her to tell me the number that called and I add > it to my dial plan that plays a recording that politely says ?go pound > sand.?**** > > ** ** > > So help me take this to the next level:**** > > **1. **What should I be returning to the caller?s switch (like temp > unavail) instead of a recording? How do I do that? > > **** > > **2. **She has an ATA and analog phone (so extra features are limited), > but I would love to be able to give her some kind of option like ?hang up > and dial *666 to add them to the list of band numbers? or maybe ?hook-flash > dial *666 and hang up? to xfer them to the dial plan that does that?**** > > ** ** > > All ideas are welcome and certainly appreciated.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/85701af0/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/85701af0/attachment.gif From msc at freeswitch.org Wed May 15 03:55:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 16:55:01 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: On Tue, May 14, 2013 at 9:38 AM, Cesar Bermudez wrote: > Hi all. > > What its the best GUI for use FS in pbx? and why? > bluebox? fusion? any other? > If you mean "FOSS GUI" then your choices really are blue.box and FusionPBX. If you want the absolute best GUI that money can buy then check out CudaTel.com. -MC > > Regards. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/14196ffc/attachment-0001.html From sos at sokhapkin.dyndns.org Wed May 15 04:08:12 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 14 May 2013 20:08:12 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: <6497345.fCohJ3gqet@sos> There is no online demo or even screenshots at . On Tuesday 14 May 2013 16:55:01 Michael Collins wrote: > On Tue, May 14, 2013 at 9:38 AM, Cesar Bermudez wrote: > > Hi all. > > > > What its the best GUI for use FS in pbx? and why? > > bluebox? fusion? any other? > > If you mean "FOSS GUI" then your choices really are blue.box and FusionPBX. > If you want the absolute best GUI that money can buy then check out > CudaTel.com. > > -MC > > > Regards. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 From gassaad at emassembly.com Wed May 15 04:27:59 2013 From: gassaad at emassembly.com (George Assaad) Date: Tue, 14 May 2013 20:27:59 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <6497345.fCohJ3gqet@sos> References: <6497345.fCohJ3gqet@sos> Message-ID: We are using FusionPBX for a year now. Excellent GUI, good support. On Tue, May 14, 2013 at 8:08 PM, Sergey Okhapkin wrote: > There is no online demo or even screenshots at . > > On Tuesday 14 May 2013 16:55:01 Michael Collins wrote: > > On Tue, May 14, 2013 at 9:38 AM, Cesar Bermudez > wrote: > > > Hi all. > > > > > > What its the best GUI for use FS in pbx? and why? > > > bluebox? fusion? any other? > > > > If you mean "FOSS GUI" then your choices really are blue.box and > FusionPBX. > > If you want the absolute best GUI that money can buy then check out > > CudaTel.com. > > > > -MC > > > > > Regards. > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/d49f29a3/attachment.html From msc at freeswitch.org Wed May 15 04:36:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 17:36:16 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <6497345.fCohJ3gqet@sos> References: <6497345.fCohJ3gqet@sos> Message-ID: On Tue, May 14, 2013 at 5:08 PM, Sergey Okhapkin wrote: > There is no online demo or even screenshots at . > We don't have a self-serve demo up at the moment. However, at cudatel.com/training there are seven videos that are way better than static screen shots. We also have a number of videos on YouTube. My favorite is Alden: http://www.youtube.com/watch?v=q9M7U8jimL4 In the meantime, if anyone is interested in learning more about the CudaTel we have a number of people available to answer questions. We also have 30-day eval units. If anyone wants to see a CudaTel in action just call us at 734-887-3000. On a future Wednesday conference call we'll do a CudaTel demo if there is sufficient interest. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/4fc79056/attachment.html From bdfoster at davri.com Wed May 15 04:37:20 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 14 May 2013 20:37:20 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: On May 14, 2013 8:01 PM, "Michael Collins" wrote: > > > > > On Tue, May 14, 2013 at 9:38 AM, Cesar Bermudez wrote: >> >> Hi all. >> >> What its the best GUI for use FS in pbx? and why? >> bluebox? fusion? any other? > > > If you mean "FOSS GUI" then your choices really are blue.box and FusionPBX. If you want the absolute best GUI that money can buy then check out CudaTel.com. > > -MC > ...best money can by for not a lot of dough. It's definitely priced right. - BDF >> >> >> Regards. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130514/dd29b036/attachment.html From sdevoy at bizfocused.com Wed May 15 05:50:21 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 14 May 2013 21:50:21 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> Message-ID: <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Thanks MC I will look at the call return in the examples. But: hash_dump all -ERR no reply From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 14, 2013 7:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In the example config there's already a "call return" feature that utilizes the last received caller ID value. If there was a valid caller ID number sent then it will be stored in the local database, which is where the call_return extension gets it from. -MC P.S. - If you haven't already started using the FS local database then check it out. Try the "hash_dump all" command at fs_cli to get an idea of what gets stored. On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: HI all, My mother has made a request and Mom's always get our best effort right? For those of you outside the US, we have a national Do Not Call Registry: www.donotcall.gov. You register your number there and in 30 days or less you are on the list. Telemarketers are required by law to check the list before calling and not call if you are on it. It does certainly help, but overseas call centers tend to ignore it and of course the excluded groups from the law "charities, political organizations, and telephone surveyors" may still call you. So it is simple enough for her to tell me the number that called and I add it to my dial plan that plays a recording that politely says "go pound sand." So help me take this to the next level: 1. What should I be returning to the caller's switch (like temp unavail) instead of a recording? How do I do that? 2. She has an ATA and analog phone (so extra features are limited), but I would love to be able to give her some kind of option like "hang up and dial *666 to add them to the list of band numbers" or maybe "hook-flash dial *666 and hang up" to xfer them to the dial plan that does that? All ideas are welcome and certainly appreciated. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/8e1d3465/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/8e1d3465/attachment-0001.gif From msc at freeswitch.org Wed May 15 06:21:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 14 May 2013 19:21:09 -0700 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: Are you using the example configs? If so you can make a local call, like to 9664, and then run hash_dump again. -MC On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: > Thanks MC I will look at the call return in the examples.**** > > ** ** > > But: hash_dump all**** > > -ERR no reply**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, May 14, 2013 7:53 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Late mother's day gift - Advanced Do > Not Call service!**** > > ** ** > > In the example config there's already a "call return" feature that > utilizes the last received caller ID value. If there was a valid caller ID > number sent then it will be stored in the local database, which is where > the call_return extension gets it from. **** > > -MC**** > > P.S. - If you haven't already started using the FS local database then > check it out. Try the "hash_dump all" command at fs_cli to get an idea of > what gets stored.**** > > ** ** > > On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: > **** > > **** > > HI all,**** > > **** > > My mother has made a request and Mom?s always get our best effort right?** > ** > > **** > > For those of you outside the US, we have a national Do Not Call Registry: > www.donotcall.gov. You register your number there and in 30 days or less > you are on the list. Telemarketers are required by law to check the list > before calling and not call if you are on it.**** > > **** > > It does certainly help, but overseas call centers tend to ignore it and of > course the excluded groups from the law ?*charities, political > organizations, and telephone surveyors*? may still call you.**** > > **** > > So it is simple enough for her to tell me the number that called and I add > it to my dial plan that plays a recording that politely says ?go pound > sand.?**** > > **** > > So help me take this to the next level:**** > > 1. What should I be returning to the caller?s switch (like temp > unavail) instead of a recording? How do I do that?**** > > 2. She has an ATA and analog phone (so extra features are limited), but > I would love to be able to give her some kind of option like ?hang up and > dial *666 to add them to the list of band numbers? or maybe ?hook-flash > dial *666 and hang up? to xfer them to the dial plan that does that?**** > > **** > > All ideas are welcome and certainly appreciated.**** > > **** > > Thanks,**** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/1a7e8664/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130514/1a7e8664/attachment.gif From alex at opensystems.net.au Wed May 15 07:43:08 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Wed, 15 May 2013 11:43:08 +0800 Subject: [Freeswitch-users] =?windows-1252?q?Error_message_=22_=5BERR=5D_s?= =?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?= =?windows-1252?q?ts=93?= In-Reply-To: References: <1339645894596-7579817.post@n2.nabble.com> <1368502318906-7590651.post@n2.nabble.com> Message-ID: It's not set to 5000 it's hitting the session limit of 1000 which is weird as our sip provider has set a limit of 100 on our trunk. It appears that some sessions aren't being closed off properly. /etc/freeswitch/autoload_configs/switch.conf.xml Max-sesssions parameter The system is a fully up to date Ubuntu server LTS install too. I was watching it last night during an active survey and it sat between 50-100 sessions. Our campaigns are set to no more than 150calls per minute. Which should keep the sessions low. Alex Ynema | IT Consultant alex at opensystems.net.au Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 15/05/2013 8:15 AM, "Alex Ynema" wrote: > It's not set to 5000 it's hitting the session limit of 1000 which is weird > as our sip provider has set a limit of 100 on our trunk. It appears that > some sessions aren't being closed off properly. > /etc/freeswitch/autoload_configs/switch.conf.xml > Max-sesssions parameter > > The system is a fully up to date Ubuntu server LTS install too. > > I was watching it last night during an active survey and it sat between > 50-100 sessions. Our campaigns are set to no more than 150calls per minute. > Which should keep the sessions low. > > Alex Ynema | IT Consultant > alex at opensystems.net.au > > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > On 15/05/2013 6:25 AM, "Steven Ayre" wrote: > >> Don't set it so high that you run out of memory. If your queue is growing >> now, with a higher limit it's going to grow more so. >> >> I suggest you look at how to speed up your consuming. Some ideas: >> >> - Use a new MySQL version - 5.5/5.6 (there will be numerous performance >> improvements over older versions) >> - Use the innodb engine (row locking this will better performance as >> multiple connections can use the table at the same time) >> - Use multiple DB connections at the same time (which will allow multiple >> DB threads therefore using all the cores of your CPU) >> - Use a large innodb buffer size (ideally this should be as large as your >> data set or at least what you actively use - this'll mean you don't need to >> read from disk as often and so reads *and* writes will be faster) >> - Use EXPLAIN to tune your queries and indexes. Don't underindex (so you >> can find/update data quickly) but also don't overindex (which will waste >> cpu and disk updating indexes that are rarely/never used). >> >> There're numerous other ways to tune it too. >> >> -Steve >> >> >> >> >> On 14 May 2013 16:43, Anthony Minessale wrote: >> >>> Its the 3rd arg to the EventConsumer constructor and it defaults to 5000 >>> >>> >>> >>> On Mon, May 13, 2013 at 10:31 PM, alex_ynema wrote: >>> >>>> Where can I set this limit in Freeswitch. >>>> I've been hitting the limit quite regularly while using it to run call >>>> campaigns. >>>> >>>> Cheers >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.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://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 >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130515/ccd58ad3/attachment-0001.html From ashish at nms.co.in Wed May 15 10:28:18 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 15 May 2013 11:58:18 +0530 Subject: [Freeswitch-users] hangup extension In-Reply-To: References: <5190AC8A.1080903@telecube.com.au> Message-ID: Hi Michael, I set in my dialplan: and in my hangup hook I am doing this: sub hupHook{ freeswitch::consoleLog("err","In Sub hupHook\n"); my $endDispo = $api->executeString("uuid_getvar $uuid endpoint_disposition XML"); chomp($endDispo);# freeswitch::consoleLog("INFO","Endpoint dispostion huphook is: $endDispo\n"); } on console its showing this: 2013-05-15 11:51:13.157662 [INFO] switch_cpp.cpp:1275 Endpoint dispostion huphook is: -ERR No such channel! On Wed, May 15, 2013 at 3:42 AM, Michael Collins wrote: > > > > On Mon, May 13, 2013 at 10:31 PM, Ashish gautam wrote: > >> Session gets dissolved in hangupHook and we cannot do anything on the >> basis of UUID > > Normally yes, but that's why we have the chan var session_in_hangup_hook. > Try setting that to true and then using your api_hangup_hook. > -MC > > >> >> On Mon, May 13, 2013 at 2:34 PM, Telecube - John wrote: >> >>> If you are using Lua script there is ' session:setHangupHook' detailed >>> here: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook >>> >>> I assume there's something similar in the xml configs >>> >>> Regards, >>> John >>> >>> >>> >>> On 13/05/13 5:20 PM, Ashish gautam wrote: >>> >>> Hi, >>> >>> Is there anything like hangup extension which gets executed after the >>> call is hung up for which the UUID is same and based on that I can do >>> things for the same call like we have in Asterisk? >>> >>> Thanks. >>> >>> Regards, >>> >>> --Ashish >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/4c7b8620/attachment.html From mehroz.ashraf85 at gmail.com Wed May 15 10:44:42 2013 From: mehroz.ashraf85 at gmail.com (Mehroz Ashraf) Date: Wed, 15 May 2013 11:44:42 +0500 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/8d5cc7d3/attachment.html From jeff at askcornerstone.net Wed May 15 11:59:59 2013 From: jeff at askcornerstone.net (Jeff Bernhardt) Date: Wed, 15 May 2013 07:59:59 +0000 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> I was going through the same thing about a month ago. I just decided to buckle down and learn it via cli. I know this isn't an option for everyone, but if you have the time to learn it, it's pretty straightforward... although I'm only doing the bare minimum with it. Jeff From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mehroz Ashraf Sent: Tuesday, May 14, 2013 8:45 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/0e62c619/attachment.html From raimund.sacherer at logitravel.com Wed May 15 12:19:03 2013 From: raimund.sacherer at logitravel.com (info) Date: Wed, 15 May 2013 10:19:03 +0200 (CEST) Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: <15534432.5814.1368605789796.JavaMail.javamailuser@localhost> Message-ID: <23490192.5864.1368605939908.JavaMail.javamailuser@localhost> On Tue, May 14, 2013 at 10:31 AM, Siri MM < sirimmfs at gmail.com > wrote: Hello All, I am facing an issue with freeswitch installed on my device. For some reason, IP address on my device sometimes seems to be 'fluctuating' between the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this change everytime, and reconfigure accordingly. However, during one such occassion, it hasn't brought up the internal profile, and isn't binding to port 5060. No other program uses this port on my device. Working log: http://pastebin.freeswitch.org/20912 Not-working log: http://pastebin.freeswitch.org/20915 Would appreciate any inputs Thanks! Hi Siri, I am a bit mystified as to how things are supposed to work if (how?) your IP get's switched between static and 127.0.0.1 (again, how?) This feels just wrong on so many levels :-) 127.0.0.1 normally is bound to the loopback interface, which, on linux, is called lo your other IP should be bound to an eth0 interface, or an internal ip and and the static IP on a router with some sort of NAT or port redirection. AFAIK linux should not allow you bind 127.0.0.1 to eth0 if it is allready bound to lo, so my best guess would be that you loose ethX and you are binding FS to all available interfaces... Alas, I don't know how to answer, because even if there should be a problem bringing up the internal profile on such a "switch", my guess is that that's probably *not* the problem you should be looking to fix for right now. Especially since, if somehow the IP really get's switched to 127.0.0.1, you will be talking only to yourself and I do not see how this should create a working setup (unless you use only an internal sip/(pri|bri) card or some USB channelbanks/gprs sticks, etc. and you only will be doing robocalls or automated IVR's ... Well, sorry mate that I can't help you more, but this really is a strange szenario, best regards, Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/ae673a60/attachment-0001.html From khorsmann at gmail.com Wed May 15 12:27:52 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 15 May 2013 10:27:52 +0200 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: References: Message-ID: Hi Siri, try to disable auto-restart in mod_sofia profile and fill in the internal ip-address. 2013/5/15 Siri MM > Hey Guys, Would appreciate any inputs! Thanks. > > > On Tue, May 14, 2013 at 10:31 AM, Siri MM wrote: > >> Hello All, >> >> I am facing an issue with freeswitch installed on my device. For some >> reason, IP address on my device sometimes seems to be 'fluctuating' between >> the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this >> change everytime, and reconfigure accordingly. However, during one such >> occassion, it hasn't brought up the internal profile, and isn't binding to >> port 5060. No other program uses this port on my device. >> >> Working log: >> http://pastebin.freeswitch.org/20912 >> >> Not-working log: >> http://pastebin.freeswitch.org/20915 >> >> Would appreciate any inputs >> >> Thanks! >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Mit freundlichen Gr??en *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/d4305219/attachment.html From vbvbrj at gmail.com Wed May 15 12:33:05 2013 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 15 May 2013 11:33:05 +0300 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> References: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> Message-ID: <51934841.6010607@gmail.com> On 15.05.2013 10:59, Jeff Bernhardt wrote: > I was going through the same thing about a month ago. I just decided to > buckle down and learn it via cli. I know this isn?t an option for > everyone, but if you have the time to learn it, it?s pretty > straightforward? although I?m only doing the bare minimum with it. All I do with cli and xml editing. Its just this way how I learned FS from start. Now I need GUI to delegate some administration, like extension configuration, to others. -- Mimiko desu. From mbodbg at gmx.net Wed May 15 12:42:24 2013 From: mbodbg at gmx.net (mbo) Date: Wed, 15 May 2013 10:42:24 +0200 Subject: [Freeswitch-users] channel selection during outbound dialing In-Reply-To: References: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> Message-ID: <7CD3228A-FFE5-449A-A6C0-F491EF4C8E58@gmx.net> Thanks for the link. It seems that the wiki is not up to date here. In the code of mod_freetdm I can see that "r" for hunt going downwards starting from highest channel and "R" hunt going upwards starting from lowest channel seems to be also available. ... } else if (*argv[1] == 'r') { direction = FTDM_RR_DOWN; } else if (*argv[1] == 'R') { direction = FTDM_RR_UP; ? Thanks Markus Am 07.05.2013 um 21:41 schrieb Michael Collins : > I believe all of that is covered here: > http://wiki.freeswitch.org/wiki/FreeTDM#Dial_Plan > > The "a" means start at channel 1 and work your way up. > -MC > > > On Tue, May 7, 2013 at 10:02 AM, mbo wrote: > We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in one group. In Asterisk it is possible to specify the channel selection mode by a prefix in the dial command (Dial(Zap/g2?) - ascending sequential hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2?) - descending sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I use a dial string like: > > freetdm/mygroup/a/ > > to dial out via a group. I'm not sure for what the "a" stand for. Are there also options for the channel selection mode available like in asterisk? > > Thanks > > Markus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/4fd6af12/attachment.html From andrew at cassidywebservices.co.uk Wed May 15 13:19:53 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 15 May 2013 10:19:53 +0100 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: With regards to the first question, one of my old university lecturers plays a not in service tone. We have something similar in the uk: http://www.tpsonline.org.uk/tps/index.html On 15 May 2013 03:21, Michael Collins wrote: > Are you using the example configs? If so you can make a local call, like > to 9664, and then run hash_dump again. > -MC > > > On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: > >> Thanks MC I will look at the call return in the examples.**** >> >> ** ** >> >> But: hash_dump all**** >> >> -ERR no reply**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Tuesday, May 14, 2013 7:53 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Late mother's day gift - Advanced Do >> Not Call service!**** >> >> ** ** >> >> In the example config there's already a "call return" feature that >> utilizes the last received caller ID value. If there was a valid caller ID >> number sent then it will be stored in the local database, which is where >> the call_return extension gets it from. **** >> >> -MC**** >> >> P.S. - If you haven't already started using the FS local database then >> check it out. Try the "hash_dump all" command at fs_cli to get an idea of >> what gets stored.**** >> >> ** ** >> >> On Tue, May 14, 2013 at 3:08 PM, Sean Devoy >> wrote:**** >> >> **** >> >> HI all,**** >> >> **** >> >> My mother has made a request and Mom?s always get our best effort right?* >> *** >> >> **** >> >> For those of you outside the US, we have a national Do Not Call Registry: >> www.donotcall.gov. You register your number there and in 30 days or >> less you are on the list. Telemarketers are required by law to check the >> list before calling and not call if you are on it.**** >> >> **** >> >> It does certainly help, but overseas call centers tend to ignore it and >> of course the excluded groups from the law ?*charities, political >> organizations, and telephone surveyors*? may still call you.**** >> >> **** >> >> So it is simple enough for her to tell me the number that called and I >> add it to my dial plan that plays a recording that politely says ?go pound >> sand.?**** >> >> **** >> >> So help me take this to the next level:**** >> >> 1. What should I be returning to the caller?s switch (like temp >> unavail) instead of a recording? How do I do that?**** >> >> 2. She has an ATA and analog phone (so extra features are limited), >> but I would love to be able to give her some kind of option like ?hang up >> and dial *666 to add them to the list of band numbers? or maybe ?hook-flash >> dial *666 and hang up? to xfer them to the dial plan that does that?**** >> >> **** >> >> All ideas are welcome and certainly appreciated.**** >> >> **** >> >> Thanks,**** >> >> Sean**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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**** >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *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/20130515/4b14602d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/4b14602d/attachment-0001.gif From steveayre at gmail.com Wed May 15 14:35:36 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 15 May 2013 11:35:36 +0100 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: Yep. Unfortunately as already noted its not perfect - plenty of people ignore it (afaik illegally, call from outside the UK, or most inciduously have permission to call you because they bought the number from someone who bought the number from someone who bought the number from someone who you accidentally ticked the 'you and anyone you sell my number to can call me' tick boxes on a form 15 years ago. -Steve On Wednesday, May 15, 2013, Andrew Cassidy wrote: > With regards to the first question, one of my old university lecturers > plays a not in service tone. > > We have something similar in the uk: > http://www.tpsonline.org.uk/tps/index.html > > > On 15 May 2013 03:21, Michael Collins wrote: > > Are you using the example configs? If so you can make a local call, like > to 9664, and then run hash_dump again. > -MC > > > On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: > > Thanks MC I will look at the call return in the examples.**** > > ** ** > > But: hash_dump all**** > > -ERR no reply**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, May 14, 2013 7:53 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Late mother's day gift - Advanced Do > Not Call service!**** > > ** ** > > In the example config there's already a "call return" feature that > utilizes the last received caller ID value. If there was a valid caller ID > number sent then it will be stored in the local database, which is where > the call_return extension gets it from. **** > > -MC**** > > P.S. - If you haven't already started using the FS local database then > check it out. Try the "hash_dump all" command at fs_cli to get an idea of > what gets stored.**** > > ** ** > > On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: > **** > > **** > > HI all,**** > > **** > > My mother has made a request and Mom?s always get our best effort right?** > ** > > **** > > For those of you outside the US, we have a national Do Not Call Registry: > www.donotcall.gov. You register your number there and in 30 days or less > you are on the list. Telemarketers are required by law to check the list > before calling and not call if you are on it.**** > > **** > > It does certainly help, but overseas call centers tend to ignore it and of > course the excluded groups from the law ?*charities, political > organizations, and telephone surveyors*? may still call you.**** > > **** > > So it is simple enough for her to tell > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 > 100 960 *F > *03300 100 961 > *E * > andrew at cassidywebservices.co.uk '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/20130515/e415b72a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/e415b72a/attachment.gif From steveayre at gmail.com Wed May 15 14:40:34 2013 From: steveayre at gmail.com (Steven Ayre) Date: Wed, 15 May 2013 11:40:34 +0100 Subject: [Freeswitch-users] =?windows-1252?q?Error_message_=22_=5BERR=5D_s?= =?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?= =?windows-1252?q?ts=93?= In-Reply-To: References: <1339645894596-7579817.post@n2.nabble.com> <1368502318906-7590651.post@n2.nabble.com> Message-ID: Sessions are not queued events. They're separate limits. You can have 1 session at a time and still hit the event queue limit if you're not pulling events out. As for 1000 sessions when there's a 100 trunk limit - sounds like you're originating many more sessions than the trunk can handle. It won't block a session until there's a line available, that session will just fail (and from the sound of it instantly be replaced by you). I'd suggest 99% of your calls are probably failing. What do your logs/CDRs tell you? You need to wait until a channel is available before you start a call. -Steve On Wednesday, May 15, 2013, Alex Ynema wrote: > > It's not set to 5000 it's hitting the session limit of 1000 which is weird > as our sip provider has set a limit of 100 on our trunk. It appears that > some sessions aren't being closed off properly. > /etc/freeswitch/autoload_configs/switch.conf.xml > Max-sesssions parameter > > The system is a fully up to date Ubuntu server LTS install too. > > I was watching it last night during an active survey and it sat between > 50-100 sessions. Our campaigns are set to no more than 150calls per minute. > Which should keep the sessions low. > > Alex Ynema | IT Consultant > alex at opensystems.net.au 'alex at opensystems.net.au');> > > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > On 15/05/2013 8:15 AM, "Alex Ynema" 'cvml', 'alex at ynema.net');>> wrote: > >> It's not set to 5000 it's hitting the session limit of 1000 which is >> weird as our sip provider has set a limit of 100 on our trunk. It appears >> that some sessions aren't being closed off properly. >> /etc/freeswitch/autoload_configs/switch.conf.xml >> Max-sesssions parameter >> >> The system is a fully up to date Ubuntu server LTS install too. >> >> I was watching it last night during an active survey and it sat between >> 50-100 sessions. Our campaigns are set to no more than 150calls per minute. >> Which should keep the sessions low. >> >> Alex Ynema | IT Consultant >> alex at opensystems.net.au > 'alex at opensystems.net.au');> >> >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> On 15/05/2013 6:25 AM, "Steven Ayre" wrote: >> >> Don't set it so high that you run out of memory. If your queue is growing >> now, with a higher limit it's going to grow more so. >> >> I suggest you look at how to speed up your consuming. Some ideas: >> >> - Use a new MySQL version - 5.5/5.6 (there will be numerous performance >> improvements over older versions) >> - Use the innodb engine (row locking this will better performance as >> multiple connections can use the table at the same time) >> - Use multiple DB connections at the same time (which will allow multiple >> DB threads therefore using all the cores of your CPU) >> - Use a large innodb buffer size (ideally this should be as large as your >> data set or at least what you actively use - this'll mean you don't need to >> read from disk as often and so reads *and* writes will be faster) >> - Use EXPLAIN to tune your queries and indexes. Don't underindex (so you >> can find/update data quickly) but also don't overindex (which will waste >> cpu and disk updating indexes that are rarely/never used). >> >> There're numerous other ways to tune it too. >> >> -Steve >> >> >> >> >> On 14 May 2013 16:43, Anthony Minessale wrote: >> >> Its the 3rd arg to the EventConsumer constructor and it defaults to 5000 >> >> >> >> On Mon, May 13, 2013 at 10:31 PM, alex_ynema wrote: >> >> Where can I set this limit in Freeswitch. >> I've been hitting the limit quite regularly while using it to run call >> campaigns. >> >> Cheers >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7590651.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://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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/ >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/2f01eba1/attachment.html From andrew at cassidywebservices.co.uk Wed May 15 15:03:58 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 15 May 2013 12:03:58 +0100 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: Indeed. I work for one of the companies that buys personal data then sells it on again... On 15 May 2013 11:35, Steven Ayre wrote: > Yep. Unfortunately as already noted its not perfect - plenty of people > ignore it (afaik illegally, call from outside the UK, or most inciduously > have permission to call you because they bought the number from someone > who bought the number from someone who bought the number from someone who > you accidentally ticked the 'you and anyone you sell my number to can call > me' tick boxes on a form 15 years ago. > > -Steve > > > On Wednesday, May 15, 2013, Andrew Cassidy wrote: > >> With regards to the first question, one of my old university lecturers >> plays a not in service tone. >> >> We have something similar in the uk: >> http://www.tpsonline.org.uk/tps/index.html >> >> >> On 15 May 2013 03:21, Michael Collins wrote: >> >> Are you using the example configs? If so you can make a local call, like >> to 9664, and then run hash_dump again. >> -MC >> >> >> On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: >> >> Thanks MC I will look at the call return in the examples.**** >> >> ** ** >> >> But: hash_dump all**** >> >> -ERR no reply**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Tuesday, May 14, 2013 7:53 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Late mother's day gift - Advanced Do >> Not Call service!**** >> >> ** ** >> >> In the example config there's already a "call return" feature that >> utilizes the last received caller ID value. If there was a valid caller ID >> number sent then it will be stored in the local database, which is where >> the call_return extension gets it from. **** >> >> -MC**** >> >> P.S. - If you haven't already started using the FS local database then >> check it out. Try the "hash_dump all" command at fs_cli to get an idea of >> what gets stored.**** >> >> ** ** >> >> On Tue, May 14, 2013 at 3:08 PM, Sean Devoy >> wrote:**** >> >> **** >> >> HI all,**** >> >> **** >> >> My mother has made a request and Mom?s always get our best effort right?* >> *** >> >> **** >> >> For those of you outside the US, we have a national Do Not Call Registry: >> www.donotcall.gov. You register your number there and in 30 days or >> less you are on the list. Telemarketers are required by law to check the >> list before calling and not call if you are on it.**** >> >> **** >> >> It does certainly help, but overseas call centers tend to ignore it and >> of course the excluded groups from the law ?*charities, political >> organizations, and telephone surveyors*? may still call you.**** >> >> **** >> >> So it is simple enough for her to tell >> >> *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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *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/20130515/2c751224/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/2c751224/attachment-0001.gif From john at telecube.com.au Wed May 15 15:15:50 2013 From: john at telecube.com.au (Telecube - John) Date: Wed, 15 May 2013 21:15:50 +1000 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: <51936E66.1090907@telecube.com.au> How about setup a keypress like #666 that transfers the call into a fax tone and also writes the caller id into a db table so that every future call from that number goes straight to a fax tone... ;-) On 15/05/13 9:03 PM, Andrew Cassidy wrote: > Indeed. I work for one of the companies that buys personal data then > sells it on again... > > > On 15 May 2013 11:35, Steven Ayre > wrote: > > Yep. Unfortunately as already noted its not perfect - plenty of > people ignore it (afaik illegally, call from outside the UK, or > most inciduously have permission to call you because they bought > the number from someone who bought the number from someone who > bought the number from someone who you accidentally ticked the > 'you and anyone you sell my number to can call me' tick boxes on a > form 15 years ago. > > -Steve > > > On Wednesday, May 15, 2013, Andrew Cassidy wrote: > > With regards to the first question, one of my old university > lecturers plays a not in service tone. > > We have something similar in the uk: > http://www.tpsonline.org.uk/tps/index.html > > > On 15 May 2013 03:21, Michael Collins wrote: > > Are you using the example configs? If so you can make a > local call, like to 9664, and then run hash_dump again. > -MC > > > On Tue, May 14, 2013 at 6:50 PM, Sean Devoy > wrote: > > Thanks MC I will look at the call return in the examples. > > But: hash_dump all > > -ERR no reply > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Michael Collins > *Sent:* Tuesday, May 14, 2013 7:53 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Late mother's day > gift - Advanced Do Not Call service! > > In the example config there's already a "call return" > feature that utilizes the last received caller ID > value. If there was a valid caller ID number sent then > it will be stored in the local database, which is > where the call_return extension gets it from. > > -MC > > P.S. - If you haven't already started using the FS > local database then check it out. Try the "hash_dump > all" command at fs_cli to get an idea of what gets stored. > > On Tue, May 14, 2013 at 3:08 PM, Sean Devoy > wrote: > > HI all, > > My mother has made a request and Mom's always get our > best effort right? > > For those of you outside the US, we have a national Do > Not Call Registry: www.donotcall.gov > . You register your number > there and in 30 days or less you are on the list. > Telemarketers are required by law to check the list > before calling and not call if you are on it. > > It does certainly help, but overseas call centers tend > to ignore it and of course the excluded groups from > the law "*charities, political organizations, and > telephone surveyors*" may still call you. > > So it is simple enough for her to tell > > *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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > *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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/daf7efb5/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/daf7efb5/attachment.gif From mike at jerris.com Wed May 15 16:42:14 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 15 May 2013 08:42:14 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: Anyone interested in us setting up a web service that you can post naughty numbers too so we can share? This could be a fun project to set up. On May 15, 2013, at 7:03 AM, Andrew Cassidy wrote: > Indeed. I work for one of the companies that buys personal data then sells it on again... > > > On 15 May 2013 11:35, Steven Ayre wrote: > Yep. Unfortunately as already noted its not perfect - plenty of people ignore it (afaik illegally, call from outside the UK, or most inciduously have permission to call you because they bought the number from someone who bought the number from someone who bought the number from someone who you accidentally ticked the 'you and anyone you sell my number to can call me' tick boxes on a form 15 years ago. > > -Steve > > > On Wednesday, May 15, 2013, Andrew Cassidy wrote: > With regards to the first question, one of my old university lecturers plays a not in service tone. > > We have something similar in the uk: http://www.tpsonline.org.uk/tps/index.html > > > On 15 May 2013 03:21, Michael Collins wrote: > Are you using the example configs? If so you can make a local call, like to 9664, and then run hash_dump again. > -MC > > > On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: > Thanks MC I will look at the call return in the examples. > > > > But: hash_dump all > > -ERR no reply > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: Tuesday, May 14, 2013 7:53 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! > > > > In the example config there's already a "call return" feature that utilizes the last received caller ID value. If there was a valid caller ID number sent then it will be stored in the local database, which is where the call_return extension gets it from. > > -MC > > P.S. - If you haven't already started using the FS local database then check it out. Try the "hash_dump all" command at fs_cli to get an idea of what gets stored. > > > > On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: > > > > HI all, > > > > My mother has made a request and Mom?s always get our best effort right? > > > > For those of you outside the US, we have a national Do Not Call Registry: www.donotcall.gov. You register your number there and in 30 days or less you are on the list. Telemarketers are required by law to check the list before calling and not call if you are on it. > > > > It does certainly help, but overseas call centers tend to ignore it and of course the excluded groups from the law ?charities, political organizations, and telephone surveyors? may still call you. > > > > So it is simple enough for her to tell > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/c8c9d31d/attachment-0001.html From john at telecube.com.au Wed May 15 17:11:05 2013 From: john at telecube.com.au (Telecube - John) Date: Wed, 15 May 2013 23:11:05 +1000 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> Message-ID: <51938969.5070707@telecube.com.au> I can imagine something like that getting a bit messy, what if some smarty pants decides to poison it by randomly generating numbers and sending them to your list? - John On 15/05/13 10:42 PM, Michael Jerris wrote: > Anyone interested in us setting up a web service that you can post > naughty numbers too so we can share? This could be a fun project to > set up. > > On May 15, 2013, at 7:03 AM, Andrew Cassidy > > wrote: > >> Indeed. I work for one of the companies that buys personal data then >> sells it on again... >> >> >> On 15 May 2013 11:35, Steven Ayre > > wrote: >> >> Yep. Unfortunately as already noted its not perfect - plenty of >> people ignore it (afaik illegally, call from outside the UK, or >> most inciduously have permission to call you because they bought >> the number from someone who bought the number from someone who >> bought the number from someone who you accidentally ticked the >> 'you and anyone you sell my number to can call me' tick boxes on >> a form 15 years ago. >> >> -Steve >> >> >> On Wednesday, May 15, 2013, Andrew Cassidy wrote: >> >> With regards to the first question, one of my old university >> lecturers plays a not in service tone. >> >> We have something similar in the uk: >> http://www.tpsonline.org.uk/tps/index.html >> >> >> On 15 May 2013 03:21, Michael Collins wrote: >> >> Are you using the example configs? If so you can make a >> local call, like to 9664, and then run hash_dump again. >> -MC >> >> >> On Tue, May 14, 2013 at 6:50 PM, Sean Devoy >> wrote: >> >> Thanks MC I will look at the call return in the examples. >> >> But: hash_dump all >> >> -ERR no reply >> >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On >> Behalf Of *Michael Collins >> *Sent:* Tuesday, May 14, 2013 7:53 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Late mother's day >> gift - Advanced Do Not Call service! >> >> In the example config there's already a "call return" >> feature that utilizes the last received caller ID >> value. If there was a valid caller ID number sent >> then it will be stored in the local database, which >> is where the call_return extension gets it from. >> >> -MC >> >> P.S. - If you haven't already started using the FS >> local database then check it out. Try the "hash_dump >> all" command at fs_cli to get an idea of what gets >> stored. >> >> On Tue, May 14, 2013 at 3:08 PM, Sean Devoy >> wrote: >> >> >> >> HI all, >> >> My mother has made a request and Mom's always get our >> best effort right? >> >> For those of you outside the US, we have a national >> Do Not Call Registry: www.donotcall.gov >> . You register your >> number there and in 30 days or less you are on the >> list. Telemarketers are required by law to check the >> list before calling and not call if you are on it. >> >> It does certainly help, but overseas call centers >> tend to ignore it and of course the excluded groups >> from the law "*charities, political organizations, >> and telephone surveyors*" may still call you. >> >> So it is simple enough for her to tell >> >> *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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> *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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/429a388b/attachment-0001.html From acrow at integrafin.co.uk Wed May 15 17:14:24 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 15 May 2013 14:14:24 +0100 Subject: [Freeswitch-users] Added to wiki - autoprovisioning via LDAP and auto queue creation example Message-ID: <51938A30.8090305@integrafin.co.uk> Hi all, I've added the following to the wiki: https://wiki.freeswitch.org/wiki/Quick_and_nasty_autoprovisioning_and_dynamic_directories_and_queues_for_Snom_and_Polycom It covers my tests in setting up provisioning for Polycom and Snom phones using and LDAP server, and some work I've done in having callers directed to a regional mod_callcenter queue based on their CLI as stored in a customer database. Coding is pretty awful, so you've been warned! Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc. 29 Clement's Lane, London EC4N 7AE. Tel: (020) 7608 4900 Fax: (020) 7608 5300. (Registered office: as above; Registered in England and Wales under number: 3727592). Authorised and regulated by the Financial Conduct Authority (entered on the Financial Services Register; no. 190856). From john at telecube.com.au Wed May 15 17:16:41 2013 From: john at telecube.com.au (Telecube - John) Date: Wed, 15 May 2013 23:16:41 +1000 Subject: [Freeswitch-users] SNMP monitoring Message-ID: <51938AB9.5000902@telecube.com.au> Does anyone have any tips or snippets for integrating Freeswitch with Cactus monitoring via SNMP that they would be prepared to share? -- Regards, John Matich Telecube Pty Ltd www.telecube.com.au Ph: 13CUBE (132823) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/53cd6aa2/attachment.html From cesar.bermudez at gmail.com Wed May 15 18:09:56 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 15 May 2013 08:09:56 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Can give me some sample on wha you tried to do about the dialplan and find that? the dialplan hidden thing. What are the differences that you spoted in the bluebox gui? Regards and thx. On Wed, May 15, 2013 at 12:44 AM, Mehroz Ashraf wrote: > About FusionPBX, i guess this is not a complete opensource solution, the > dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is good > and support is alive. > > and yes, bluebox seems to be dead now, howeverm very nice interface, > completely opensource. i am having some issues in its customization and > there seems to be no support alive! > > also note that, the iso image has different bluebox interface and git > version has different! and no one to help! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/db7f2f45/attachment.html From cesar.bermudez at gmail.com Wed May 15 18:11:01 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 15 May 2013 08:11:01 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> References: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> Message-ID: Yes i know the cli and conf are the most powerfull tools, but if you have to delegate things, not all the people can do that, so for that reason i need the GUI. Regards. On Wed, May 15, 2013 at 1:59 AM, Jeff Bernhardt wrote: > I was going through the same thing about a month ago. I just decided to > buckle down and learn it via cli. I know this isn?t an option for everyone, > but if you have the time to learn it, it?s pretty straightforward? although > I?m only doing the bare minimum with it.**** > > ** ** > > Jeff **** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mehroz > Ashraf > *Sent:* Tuesday, May 14, 2013 8:45 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Recommendation for GUI to PBX**** > > ** ** > > About FusionPBX, i guess this is not a complete opensource solution, the > dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is good > and support is alive.**** > > ** ** > > and yes, bluebox seems to be dead now, howeverm very nice interface, > completely opensource. i am having some issues in its customization and > there seems to be no support alive!**** > > ** ** > > also note that, the iso image has different bluebox interface and git > version has different! and no one to help!**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/a515da17/attachment.html From cesar.bermudez at gmail.com Wed May 15 18:12:24 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 15 May 2013 08:12:24 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <51934841.6010607@gmail.com> References: <8A9716A5B256904FB1F07C050F9CCCCB020CE9F6@mail2.firstdataworks.net> <51934841.6010607@gmail.com> Message-ID: I m in the same path, delegate the administration of the pbx to others, but they cannot use the cli and the best thing to do for these is use a GUI, and start from scratch a GUI for now its not a option ( time ) . Regards. On Wed, May 15, 2013 at 2:33 AM, Mimiko wrote: > On 15.05.2013 10:59, Jeff Bernhardt wrote: > > I was going through the same thing about a month ago. I just decided to > > buckle down and learn it via cli. I know this isn?t an option for > > everyone, but if you have the time to learn it, it?s pretty > > straightforward? although I?m only doing the bare minimum with it. > > All I do with cli and xml editing. Its just this way how I learned FS > from start. Now I need GUI to delegate some administration, like > extension configuration, to others. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/0d9e2da6/attachment.html From brian at freeswitch.org Wed May 15 18:26:30 2013 From: brian at freeswitch.org (Brian West) Date: Wed, 15 May 2013 09:26:30 -0500 Subject: [Freeswitch-users] Polycom phones Message-ID: Anyone have a good cheap/donation I could acquire these Polycom devices IP320,IP321,IP331,IP450,IP4000? Possibly an ip7000 and Soundstation Duo at deeply discounted prices? I've already purchased an IP301, IP430, IP600 and IP601, they should be here friday... if anyone has the above devices and would like to donate/sell cheap for my testing lab please email me off list, I'm trying to round out my Polycom test set so I can begin the process of pushing 4.x for the models that support it and verify that we don't have any weird interactions for the ones left behind in 3.1.8 and 3.2.7 .. Thanks, /b From mburakbor at gmail.com Wed May 15 18:46:38 2013 From: mburakbor at gmail.com (=?ISO-8859-9?Q?Burak_BorYaz=FDl=FDm?=) Date: Wed, 15 May 2013 17:46:38 +0300 Subject: [Freeswitch-users] Freeswitch User Adding In-Reply-To: References: <15f401ce4e5c$dae01780$90a04680$@bizfocused.com> Message-ID: log in pastebin: http://pastebin.freeswitch.com/20919 thanks in advance... 2013/5/13 Burak BorYaz?l?m > @Sean what do you mean with crypto stuff? If you mean tls and srtp, I > didnt use them while testing. I am still trying to solve what is changing > when the number digit length is changed? > > @Brian yes an example will be helpful if you have time for it, I will be > very pleased. > > > 2013/5/12 Brian Foster > >> This is exactly why we use the user_exists call. No matter what the >> user's id is it works. I don't know why that isn't in the default dialplan >> but if you need an example I can give you one when I get home. >> >> -BDF >> On May 11, 2013 7:07 PM, "Sean Devoy" wrote: >> >>> I cannot tell you why this is only a problem with extensions w/ 5 >>> digits, but I can tell you what failed here. Your underlying error is:* >>> *** >>> >>> 2013-05-11 12:14:14.980444 [ERR] sofia_glue.c:4927 a=crypto in RTP/AVP, >>> refer to rfc3711**** >>> >>> 2013-05-11 12:14:14.980444 [ERR] mod_sofia.c:2789 CODEC NEGOTIATION >>> ERROR. SDP: >>> **** >>> >>> v=0 >>> **** >>> >>> o=- 3577252345 3577252345 IN IP4 >>> 141.196.174.60 >>> **** >>> >>> s=pjmedia >>> **** >>> >>> c=IN IP4 >>> 141.196.174.60 >>> **** >>> >>> t=0 0 >>> >>> **** >>> >>> m=audio 4010 RTP/AVP 8 0 3 >>> 101 >>> *** >>> * >>> >>> c=IN IP4 >>> 141.196.174.60 >>> **** >>> >>> a=rtpmap:8 >>> PCMA/8000 >>> **** >>> >>> a=rtpmap:0 >>> PCMU/8000 >>> >>> **** >>> >>> a=rtpmap:3 >>> GSM/8000 >>> **** >>> >>> a=rtpmap:101 >>> telephone-event/8000 >>> **** >>> >>> a=fmtp:101 >>> 0-15 >>> **** >>> >>> a=rtcp:4011 IN IP4 >>> 192.168.43.10 >>> >>> **** >>> >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >>> **** >>> >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >>> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >>> **** >>> >>> ** ** >>> >>> 2013-05-11 12:14:14.980444 [DEBUG] switch_core_session.c:830 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK]**** >>> >>> 2013-05-11 12:14:14.980444 [DEBUG] switch_channel.c:2994 >>> (sofia/internal/60021 at my.server.ip.address) Callstate Change RINGING -> >>> HANGUP **** >>> >>> 2013-05-11 12:14:14.980444 [NOTICE] switch_channel.c:3216 Hangup >>> sofia/internal/60021 at my.server.ip.address [CS_EXECUTE] >>> [INCOMPATIBLE_DESTINATION]**** >>> >>> ** ** >>> >>> Can you try it without the crypto stuff and paste the same output to >>> pastebin.freeswitch.org (not here in email)?**** >>> >>> ** ** >>> >>> ** ** >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Burak >>> BorYazilim >>> *Sent:* Saturday, May 11, 2013 7:28 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Freeswitch User Adding**** >>> >>> ** ** >>> >>> First off all thanks for your kind helps.**** >>> >>> ** ** >>> >>> Let me tell you the situation more clearly.**** >>> >>> ** ** >>> >>> Firstly I used Android phones as sip clients with the open source sip >>> client program, CSipSimple. My server computer has Ubuntu 12.04 LTS with >>> one static ip address.**** >>> >>> freeswitch version output: FreeSWITCH Version >>> 1.3.13b+git~20130205T003128Z~70a9560306 (git 70a9560 2013-02-05 00:31:28Z) >>> **** >>> >>> We were using the system with 4 digits number sip account names without >>> any problem. Also the account name will be used as dial number. An example >>> of the user xml file is below. I changed the regex in dialpan/default.xml. >>> The changings were perfectly succesfull with, again, 4 digits number. But >>> when trying to include any other number of digits (3, 5 and 6 were tested), >>> it doesn't work(When I change the regex to accept only 5 digits numbers, 4 >>> digits ones didnt work as expected). I really could not understand why it >>> is happening. Why there was no problem with 4 digits number and why the >>> exact same system does not work with this basic change.**** >>> >>> ** ** >>> >>> To explain the errors more, I want to talk about my tests. Firstly I >>> used 4 digits number user. This test repeated with tls and srtp. These two >>> test were succesfull. **** >>> >>> And the secand test is same with first test but with 5 digits numbers >>> without tls and srtp. Registration was succesfull but cant call. (or the >>> call could not be forwarded) Of course I changed the regex and execute >>> reloadxml in this test. GSM (8kHz) and G722(16kHz) codecs were used. As you >>> see below, CODEC NEGOTIATION ERROR occured. After getting this error I >>> changed the codes as SILK(16kHz). With this change I made the third test. >>> Third test result is also below. **** >>> >>> ** ** >>> >>> show registretions output: **** >>> >>> >>> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname,metadata >>> **** >>> >>> >>> 60022,my.server.ip.address,oOnxiVJFAgQWLABrD01UsYTVOY3TVSlx,sofia/internal/sip:60022 at 141.196.174.60:52245 >>> ;ob,1368271435,141.196.174.60,52245,udp,server,**** >>> >>> >>> 60021,my.server.ip.address,5r4MTDiZPhs5qzdin9A3hEUh1zZsdqqk,sofia/internal/sip:60021 at 141.196.174.60:57938 >>> ;ob,1368271446,141.196.174.60,57938,udp,server,**** >>> >>> ** ** >>> >>> ** ** >>> >>> @Sean, sorry but I cant post my whole log file because my server has got >>> a network attack so I am quite busy with this attacker. And also I dont >>> think CSipSimple has a problem with 5 digits because this system (with 5 >>> digits) was succesfull with Kamailio and CSipSimple.**** >>> >>> ** ** >>> >>> If I could not be clear please let me know. Thanks...**** >>> >>> ** ** >>> >>> user xml example: This was exact same with the default users in >>> freeswitch. Only the number changed.**** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> >>> **** >>> >>> **** >>> >>> >> value="$${outbound_caller_name}"/>**** >>> >>> >> value="$${outbound_caller_id}"/>**** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> **** >>> >>> ** ** >>> >>> ** ** >>> >>> ** ** >>> >>> The second test error:**** >>> >>> ** ** >>> >>> 2013-05-11 12:13:56.800443 [DEBUG] sofia_reg.c:1511 Send challenge for >>> [60021 at my.server.ip.address]**** >>> >>> 2013-05-11 12:13:56.900444 [DEBUG] sofia_reg.c:1511 Send challenge for >>> [60021 at my.server.ip.address] >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'record_stereo' = 'true' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_gateway' = 'example.com' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_areacode' = '918' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'transfer_fallback_extension' = 'operator' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'toll_allow' = 'domestic,international,local' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'accountcode' = '60021' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'user_context' = 'default' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_name' = 'Extension 60021' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_number' = '60021' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_name' = 'FreeSWITCH' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_number' = '0000000000' >>> **** >>> >>> 2013-05-11 12:13:56.920444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'callgroup' = 'techsupport' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'record_stereo' = 'true' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_gateway' = 'example.com' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_areacode' = '918' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'transfer_fallback_extension' = 'operator' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'toll_allow' = 'domestic,international,local' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'accountcode' = '60021' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'user_context' = 'default' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_name' = 'Extension 60021' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_number' = '60021' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_name' = 'FreeSWITCH' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_number' = '0000000000' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'callgroup' = 'techsupport' >>> **** >>> >>> 2013-05-11 12:13:56.980451 [DEBUG] sofia_reg.c:1683 Register: >>> >>> **** >>> >>> From: [60021 at my.server.ip.address] >>> >>> **** >>> >>> Contact: ["user" ] >>> >>> **** >>> >>> Expires: [900] >>> >>> **** >>> >>> 2013-05-11 12:14:06.820443 [DEBUG] sofia_reg.c:1511 Send challenge for >>> [60022 at my.server.ip.address] >>> **** >>> >>> 2013-05-11 12:14:06.900444 [DEBUG] sofia_reg.c:1511 Send challenge for >>> [60022 at my.server.ip.address] >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'record_stereo' = 'true' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_gateway' = 'example.com' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_areacode' = '918' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'transfer_fallback_extension' = 'operator' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'toll_allow' = 'domestic,international,local' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'accountcode' = '60022' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'user_context' = 'default' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_name' = 'Extension 60022' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_number' = '60022' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_name' = 'FreeSWITCH' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_number' = '0000000000' >>> **** >>> >>> 2013-05-11 12:14:06.980444 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'callgroup' = 'techsupport' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'record_stereo' = 'true' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_gateway' = 'example.com' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_areacode' = '918' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'transfer_fallback_extension' = 'operator' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'toll_allow' = 'domestic,international,local' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'accountcode' = '60022' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'user_context' = 'default' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_name' = 'Extension 60022' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_number' = '60022' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_name' = 'FreeSWITCH' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_number' = '0000000000' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'callgroup' = 'techsupport' >>> **** >>> >>> 2013-05-11 12:14:07.020442 [DEBUG] sofia_reg.c:1683 Register: >>> >>> **** >>> >>> From: [60022 at my.server.ip.address] >>> >>> **** >>> >>> Contact: ["user" ] >>> >>> **** >>> >>> Expires: [900] >>> >>> **** >>> >>> 2013-05-11 12:14:14.640437 [NOTICE] switch_channel.c:968 New Channel >>> sofia/internal/60021 at my.server.ip.address[26730cd2-ba1b-11e2-acc5-bda7cbfd9554] >>> **** >>> >>> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK]**** >>> >>> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:415 ( >>> sofia/internal/60021 at my.server.ip.address) Running State Change CS_NEW >>> **** >>> >>> 2013-05-11 12:14:14.640437 [DEBUG] switch_core_state_machine.c:433 ( >>> sofia/internal/60021 at my.server.ip.address) State NEW >>> **** >>> >>> 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:7733 IP 141.196.174.60 >>> Rejected by acl "domains". Falling back to Digest auth. >>> **** >>> >>> 2013-05-11 12:14:14.660438 [DEBUG] sofia_reg.c:1511 Send challenge for >>> [60022 at my.server.ip.address] >>> **** >>> >>> 2013-05-11 12:14:14.660438 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.660438 [DEBUG] sofia.c:1719 detaching session >>> 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 >>> **** >>> >>> 2013-05-11 12:14:14.780439 [DEBUG] sofia.c:1811 Re-attaching to session >>> 26730cd2-ba1b-11e2-acc5-bda7cbfd9554 >>> **** >>> >>> 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.780439 [DEBUG] switch_core_session.c:975 Send signal >>> sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:7733 IP 141.196.174.60 >>> Rejected by acl "domains". Falling back to Digest auth. >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'record_stereo' = 'true' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_gateway' = 'example.com' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'default_areacode' = '918' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'transfer_fallback_extension' = 'operator' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'toll_allow' = 'domestic,international,local' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'accountcode' = '60021' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'user_context' = 'default' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_name' = 'Extension 60021' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'effective_caller_id_number' = '60021' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_name' = 'FreeSWITCH' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'outbound_caller_id_number' = '0000000000' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia_reg.c:2767 event_add_header -> >>> 'callgroup' = 'techsupport' >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5578 Channel >>> sofia/internal/60021 at my.server.ip.address entering state >>> [received][100] **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5589 Remote SDP: >>> >>> **** >>> >>> v=0 >>> >>> **** >>> >>> o=- 3577252345 3577252345 IN IP4 141.196.174.60 >>> >>> **** >>> >>> s=pjmedia >>> >>> **** >>> >>> c=IN IP4 141.196.174.60 >>> >>> **** >>> >>> t=0 0 >>> >>> **** >>> >>> m=audio 4010 RTP/AVP 8 0 3 101 >>> >>> **** >>> >>> c=IN IP4 141.196.174.60 >>> >>> **** >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> **** >>> >>> a=rtpmap:0 PCMU/8000 >>> >>> **** >>> >>> a=rtpmap:3 GSM/8000 >>> >>> **** >>> >>> a=rtpmap:101 telephone-event/8000 >>> >>> **** >>> >>> a=fmtp:101 0-15 >>> >>> **** >>> >>> a=rtcp:4011 IN IP4 192.168.43.10 >>> >>> **** >>> >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >>> **** >>> >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >>> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >>> **** >>> >>> ** ** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] sofia.c:5802 ( >>> sofia/internal/60021 at my.server.ip.address) State Change CS_NEW -> >>> CS_INIT **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send >>> signal sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( >>> sofia/internal/60021 at my.server.ip.address) Running State Change CS_INIT >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( >>> sofia/internal/60021 at my.server.ip.address) State INIT >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:86 >>> sofia/internal/60021 at my.server.ip.address SOFIA INIT >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:126 ( >>> sofia/internal/60021 at my.server.ip.address) State Change CS_INIT -> >>> CS_ROUTING **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_session.c:1291 Send >>> signal sofia/internal/60021 at my.server.ip.address [BREAK] >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:454 ( >>> sofia/internal/60021 at my.server.ip.address) State INIT going to sleep >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:415 ( >>> sofia/internal/60021 at my.server.ip.address) Running State Change >>> CS_ROUTING **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_channel.c:2003 ( >>> sofia/internal/60021 at my.server.ip.address) Callstate Change DOWN -> >>> RINGING **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:470 ( >>> sofia/internal/60021 at my.server.ip.address) State ROUTING >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] mod_sofia.c:149 >>> sofia/internal/60021 at my.server.ip.address SOFIA ROUTING >>> **** >>> >>> 2013-05-11 12:14:14.800439 [DEBUG] switch_core_state_machine.c:117 >>> sofia/internal/60021 at my.server.ip.address Standard ROUTING >>> **** >>> >>> 2013-05-11 12:14:14.800439 [INFO] mod_dialplan_xml.c:557 Processing >>> 60021 <60021>->60022 in context default**** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->unloop] continue=false**** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >>> [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->tod_example] continue=true >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch >>> (FAIL) [tod_example] break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->holiday_example] continue=true >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Date/TimeMatch >>> (FAIL) [holiday_example] break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->global-intercept] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [global-intercept] destination_number(60022) =~ /^886$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->group-intercept] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [group-intercept] destination_number(60022) =~ /^\*8$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->intercept-ext] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [intercept-ext] destination_number(60022) =~ /^\*\*(\d+)$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->redial] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [redial] destination_number(60022) =~ /^(redial|870)$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->global] continue=true >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [global] ${call_debug}(false) =~ /^true$/ break=never >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [global] ${sip_has_crypto}() =~ >>> /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >>> [global] ${endpoint_disposition}(DELAYED NEGOTIATION) =~ /^(DELAYED >>> NEGOTIATION)/ break=on-false **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >>> [global] ${switch_r_sdp}(v=0 >>> **** >>> >>> o=- 3577252345 3577252345 IN IP4 141.196.174.60 >>> >>> **** >>> >>> s=pjmedia >>> >>> **** >>> >>> c=IN IP4 141.196.174.60 >>> >>> **** >>> >>> t=0 0 >>> >>> **** >>> >>> m=audio 4010 RTP/AVP 8 0 3 101 >>> >>> **** >>> >>> c=IN IP4 141.196.174.60 >>> >>> **** >>> >>> a=rtpmap:8 PCMA/8000 >>> >>> **** >>> >>> a=rtpmap:0 PCMU/8000 >>> >>> **** >>> >>> a=rtpmap:3 GSM/8000 >>> >>> **** >>> >>> a=rtpmap:101 telephone-event/8000 >>> >>> **** >>> >>> a=fmtp:101 0-15 >>> >>> **** >>> >>> a=rtcp:4011 IN IP4 192.168.43.10 >>> >>> **** >>> >>> a=crypto:1 AES_CM_128_HMAC_SHA1_80 >>> inline:fig56WojEoKmN07gnvdJZ9Mk6lznskMJszpBOqik >>> **** >>> >>> a=crypto:2 AES_CM_128_HMAC_SHA1_32 >>> inline:HUiy486/260zwSkQ0Z771fKC+g48P9cYEXNqlEYO >>> **** >>> >>> ) =~ /(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)/ break=never >>> >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(sip_secure_media=true) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> export(sip_secure_media=true) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Absolute Condition >>> [global] >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-last_dial/global/${uuid}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->snom-demo-2] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [snom-demo-2] destination_number(60022) =~ /^9001$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->snom-demo-1] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [snom-demo-1] destination_number(60022) =~ /^9000$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->eavesdrop] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [eavesdrop] destination_number(60022) =~ /^88(\d{4})$|^\*0(.*)$/ >>> break=on-false **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->eavesdrop] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [eavesdrop] destination_number(60022) =~ /^779$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->call_return] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [call_return] destination_number(60022) =~ /^\*69$|^869$|^lcr$/ >>> break=on-false **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->del-group] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [del-group] destination_number(60022) =~ /^80(\d{2})$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->add-group] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [add-group] destination_number(60022) =~ /^81(\d{2})$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->call-group-simo] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [call-group-simo] destination_number(60022) =~ /^82(\d{2})$/ break=on-false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->call-group-order] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [call-group-order] destination_number(60022) =~ /^83(\d{2})$/ >>> break=on-false **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->extension-intercom] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (FAIL) >>> [extension-intercom] destination_number(60022) =~ /^8(10[01][0-9])$/ >>> break=on-false **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address parsing >>> [default->Local_Extension] continue=false >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Regex (PASS) >>> [Local_Extension] destination_number(60022) =~ >>> /^([0-9][0-9][0-9][0-9]|[0-9][0-9][0-9][0-9][0-9])$/ break=on-false >>> >>> >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> export(dialed_extension=60022) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> bind_meta_app(1 b s execute_extension::dx XML features) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> bind_meta_app(2 b s >>> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) >>> >>> >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> bind_meta_app(3 b s execute_extension::cf XML features) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> bind_meta_app(4 b s execute_extension::att_xfer XML features) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(ringback=${us-ring}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(transfer_ringback=local_stream://moh) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(call_timeout=30) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(hangup_after_bridge=true) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(continue_on_fail=true) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} >>> var callgroup)}) **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) >>> **** >>> >>> Dialplan: sofia/internal/60021 at my.server.ip.address Action >>> hash(insert/${domain_name}-last_dial_ext/global/${uuid}) >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> ... >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130515/73c48fd0/attachment-0001.html From jmesquita at freeswitch.org Wed May 15 19:00:29 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 15 May 2013 12:00:29 -0300 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <519187A9.2040102@fidelityvoice.com> References: <51916519.3030203@fidelityvoice.com> <51916729.3020501@quentustech.com> <519187A9.2040102@fidelityvoice.com> Message-ID: Congrats to everyone involved!!!! Jo?o Mesquita FreeSWITCH? Solutions On Mon, May 13, 2013 at 9:39 PM, Jeff Pyle wrote: > My standard FS install on Debian includes the following packages: > > freeswitch > freeswitch-init > freeswitch-mod-bv > freeswitch-mod-codec2 > freeswitch-mod-commands > freeswitch-mod-console > freeswitch-mod-dialplan-xml > freeswitch-mod-dptools > freeswitch-mod-event-socket > freeswitch-mod-g729 > freeswitch-mod-hash > freeswitch-mod-ilbc > freeswitch-mod-logfile > freeswitch-mod-opus > freeswitch-mod-silk > freeswitch-mod-siren > freeswitch-mod-sofia > freeswitch-mod-spandsp > freeswitch-mod-speex > freeswitch-mod-syslog > freeswitch-mod-timerfd > freeswitch-systemd > freeswitch-sysvinit > > I noticed this thread mentioning 1.2.9 was available, so I decided to move > from my custom repo from HEAD back to the one at files.freeswitch.org. > And then I noticed these two packages weren't available. > > > - Jeff > > > > William King wrote: > > Which meta package did you try to install? > > 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 05/13/2013 03:11 PM, Jeff Pyle wrote: > > Same situation here? apt-get complaining: > > Package freeswitch-mod-ilbc is not available, but is referred to by > another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > Package freeswitch-mod-siren is not available, but is referred to by > another package. > This may mean that the package is missing, has been obsoleted, or > is only available from another source > > E: Package 'freeswitch-mod-siren' has no installation candidate > E: Package 'freeswitch-mod-ilbc' has no installation candidate > > Just give it a shot later? > > > - Jeff > > > Ken Rice wrote: > > Depends on which mirror on the CDN you hit... They can be slow at > updating sometimes > > On 5/13/13 4:30 PM, "Steven Ayre" wrote: > > Ah ok. I only saw squeeze and sid just now, but I see wheezy there > now. Perhaps it was an out of date mirror? :) > > > > On 13 May 2013 22:25, Ken Rice wrote: > > There are packages there for Squeeze, Wheezy and Sid... Jesse > packages are in the early stages > > > > On 5/13/13 4:18 PM, "Steven Ayre" > wrote: > > Great news! Any idea when Wheezy packages will be added? > > -Steve > > > On 13 May 2013 21:52, Ken Rice > wrote: > > Hey Guys, > > Just a quick heads up, As you know last week we > released FreeSWITCH 1.2.9 and I said packages were > coming. > > We now have both YUM and Debian Repo?s for you guys to > install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the > deb?s and rpm?s for it > > > -- > Ken > _http://www.FreeSWITCH.orghttp://www.ClueCon.comhttp://www.OSTAG.org > _irc.freenode.net #freeswitch > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130515/1b1bc60c/attachment.html From cal.leeming at simplicitymedialtd.co.uk Wed May 15 19:45:10 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 15 May 2013 16:45:10 +0100 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: Nice timing, Google IO is today :( Cal On Mon, May 13, 2013 at 9:52 PM, Ken Rice wrote: > Hey Guys, > > Just a quick heads up, As you know last week we released FreeSWITCH 1.2.9 > and I said packages were coming. > > We now have both YUM and Debian Repo?s for you guys to install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and rpm?s for > it > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/9b034f07/attachment.html From bpriddy at bryantschools.org Wed May 15 19:48:39 2013 From: bpriddy at bryantschools.org (Blake Priddy) Date: Wed, 15 May 2013 10:48:39 -0500 Subject: [Freeswitch-users] Decline Message-ID: Ok I talked to you guys previously about this and our gateway told me I am sending the call to them incorrectly.. Which it is just 2 phones that are doing this. The rest are fine? http://pastebin.com/3ppLaatg Any thing in there throw a flag? -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/0b39d5e8/attachment-0001.html From mikemol at gmail.com Wed May 15 19:56:40 2013 From: mikemol at gmail.com (Michael Mol) Date: Wed, 15 May 2013 11:56:40 -0400 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: <5193B038.8000202@gmail.com> I only joined this list today...but I'll note I ran into a problem with the YUM-based installation resulting in a freeswitch service which would die with SIGABRT (according to strace) when I tried setting it and Bluebox up on a CentOS 6.4 machine. Compiling from source does not have this issue for me. On 05/15/2013 11:45 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Nice timing, Google IO is today :( > > Cal > > On Mon, May 13, 2013 at 9:52 PM, Ken Rice > wrote: > > Hey Guys, > > Just a quick heads up, As you know last week we released FreeSWITCH > 1.2.9 and I said packages were coming. > > We now have both YUM and Debian Repo?s for you guys to install from... > > For Debian check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > For Centos check out > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > These will get you the stable branch for now... > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and > rpm?s for it > > > -- > Ken > _http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > _irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/101481fb/attachment.bin From cal.leeming at simplicitymedialtd.co.uk Wed May 15 20:07:05 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 15 May 2013 17:07:05 +0100 Subject: [Freeswitch-users] JIRA contributions - 15/05/2013 Message-ID: Hello all, Last week I made some contributions on JIRA by triaging some tickets. FS-3964 - requested larger bounty offer, if no bite then reclassify as nice to have FS-4211 - requested further info, close after 2 weeks if no response. FS-4281 - requested more info, waiting on core devs FS-4891 - requested more info, updated wiki docs, close after 2 weeks if no response FS-5154 - requested OP create new ticket for seperate issues, close after 2 weeks if no response However, I was unable to action them due to lack of access on JIRA. I sent an email to Ken/Michael requesting access to close tickets and mark as resolved etc, but realised this was not the appropriate way as you shouldn't need to have extra privileges in order to do ticket triage. Therefore, I'd like to open up discussion about how we can improve JIRA to allow for public contributions and make ticket triaging easier. My suggestions are; * Add a new step in the workflow that switches it into "needs more client info", so it automatically closes the ticket after X days if no extra info is received. * Have a "maybe one day/needs contributions" status in the workflow, rather than leaving the ticket as open.. that way we can see really how many are waiting, and how many are waiting on contributions. * Have a "needs design decision" status in the workflow, indicating that a core dev needs to make the final call * Introduce an official 5-for-1 ticket system where users can triage 5 tickets and request that a core dev look at any ticket of their choice in return. I think we stand a much better chance of getting a lot more public contributions if we implement these changes. Probably a bit too short notice to discuss on todays conf call, but it would be good to get some ideas/feedback in time for next weeks call. Modifying JIRA and making workflow changes is no easy task, so it would need to be thoroughly discussed and understood. I'll be on the call today 20 minutes early today if anyone wants to discuss. Any thoughts/ideas? Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/240ddf3c/attachment.html From krice at freeswitch.org Wed May 15 20:15:15 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 15 May 2013 11:15:15 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <5193B038.8000202@gmail.com> Message-ID: To figure out whats causing this we really need a ticket opened on jira with all the good information there... This will allow us to see whats going on... See http://wiki.freeswitch.org/wiki/Reporting_Bugs for more info On 5/15/13 10:56 AM, "Michael Mol" wrote: > I only joined this list today...but I'll note I ran into a problem with > the YUM-based installation resulting in a freeswitch service which would > die with SIGABRT (according to strace) when I tried setting it and > Bluebox up on a CentOS 6.4 machine. Compiling from source does not have > this issue for me. > > > On 05/15/2013 11:45 AM, Cal Leeming [Simplicity Media Ltd] wrote: >> Nice timing, Google IO is today :( >> >> Cal >> >> On Mon, May 13, 2013 at 9:52 PM, Ken Rice > > wrote: >> >> Hey Guys, >> >> Just a quick heads up, As you know last week we released FreeSWITCH >> 1.2.9 and I said packages were coming. >> >> We now have both YUM and Debian Repo?s for you guys to install from... >> >> For Debian check out >> >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packag >> es >> For Centos check out >> >> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Ins >> tallation >> >> These will get you the stable branch for now... >> >> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and >> rpm?s for it >> >> >> -- >> Ken >> _http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> _irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From cal.leeming at simplicitymedialtd.co.uk Wed May 15 20:16:07 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Wed, 15 May 2013 17:16:07 +0100 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <5193B038.8000202@gmail.com> References: <5193B038.8000202@gmail.com> Message-ID: Thank you for reporting this, and welcome to FS :) Cal On Wed, May 15, 2013 at 4:56 PM, Michael Mol wrote: > I only joined this list today...but I'll note I ran into a problem with > the YUM-based installation resulting in a freeswitch service which would > die with SIGABRT (according to strace) when I tried setting it and > Bluebox up on a CentOS 6.4 machine. Compiling from source does not have > this issue for me. > > > On 05/15/2013 11:45 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > Nice timing, Google IO is today :( > > > > Cal > > > > On Mon, May 13, 2013 at 9:52 PM, Ken Rice > > wrote: > > > > Hey Guys, > > > > Just a quick heads up, As you know last week we released FreeSWITCH > > 1.2.9 and I said packages were coming. > > > > We now have both YUM and Debian Repo?s for you guys to install > from... > > > > For Debian check out > > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packages > > For Centos check out > > > http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Installation > > > > These will get you the stable branch for now... > > > > TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and > > rpm?s for it > > > > > > -- > > Ken > > _http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > _irc.freenode.net #freeswitch > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/d595f209/attachment-0001.html From mishehu at freeswitch.org Wed May 15 20:24:22 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Wed, 15 May 2013 11:24:22 -0500 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <51938969.5070707@telecube.com.au> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> Message-ID: <5193B6B6.4080907@freeswitch.org> There are already sites like this up, but they don't act as a blacklist. More so they're forums where people talk about what type of calls they received from a given phone number. -Yossi On 05/15/2013 08:11 AM, Telecube - John wrote: > I can imagine something like that getting a bit messy, what if some > smarty pants decides to poison it by randomly generating numbers and > sending them to your list? > > - John > > On 15/05/13 10:42 PM, Michael Jerris wrote: >> Anyone interested in us setting up a web service that you can post >> naughty numbers too so we can share? This could be a fun project to >> set up. >> >> On May 15, 2013, at 7:03 AM, Andrew Cassidy >> > > wrote: >> >>> Indeed. I work for one of the companies that buys personal data then >>> sells it on again... >>> >>> >>> On 15 May 2013 11:35, Steven Ayre >> > wrote: >>> >>> Yep. Unfortunately as already noted its not perfect - plenty of >>> people ignore it (afaik illegally, call from outside the UK, or >>> most inciduously have permission to call you because they bought >>> the number from someone who bought the number from someone who >>> bought the number from someone who you accidentally ticked the >>> 'you and anyone you sell my number to can call me' tick boxes on >>> a form 15 years ago. >>> >>> -Steve >>> >>> >>> On Wednesday, May 15, 2013, Andrew Cassidy wrote: >>> >>> With regards to the first question, one of my old university >>> lecturers plays a not in service tone. >>> >>> We have something similar in the uk: >>> http://www.tpsonline.org.uk/tps/index.html >>> >>> >>> On 15 May 2013 03:21, Michael Collins >>> wrote: >>> >>> Are you using the example configs? If so you can make a >>> local call, like to 9664, and then run hash_dump again. >>> -MC >>> >>> >>> On Tue, May 14, 2013 at 6:50 PM, Sean Devoy >>> wrote: >>> >>> Thanks MC I will look at the call return in the >>> examples. >>> >>> But: hash_dump all >>> >>> -ERR no reply >>> >>> *From:*freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] >>> *On Behalf Of *Michael Collins >>> *Sent:* Tuesday, May 14, 2013 7:53 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* Re: [Freeswitch-users] Late mother's day >>> gift - Advanced Do Not Call service! >>> >>> In the example config there's already a "call >>> return" feature that utilizes the last received >>> caller ID value. If there was a valid caller ID >>> number sent then it will be stored in the local >>> database, which is where the call_return extension >>> gets it from. >>> >>> -MC >>> >>> P.S. - If you haven't already started using the FS >>> local database then check it out. Try the "hash_dump >>> all" command at fs_cli to get an idea of what gets >>> stored. >>> >>> On Tue, May 14, 2013 at 3:08 PM, Sean Devoy >>> wrote: >>> >>> >>> >>> HI all, >>> >>> My mother has made a request and Mom's always get >>> our best effort right? >>> >>> For those of you outside the US, we have a national >>> Do Not Call Registry: www.donotcall.gov >>> . You register your >>> number there and in 30 days or less you are on the >>> list. Telemarketers are required by law to check the >>> list before calling and not call if you are on it. >>> >>> It does certainly help, but overseas call centers >>> tend to ignore it and of course the excluded groups >>> from the law "*charities, political organizations, >>> and telephone surveyors*" may still call you. >>> >>> So it is simple enough for her to tell >>> >>> *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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> >>> -- >>> *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 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/fb41e008/attachment-0001.html From mikemol at gmail.com Wed May 15 20:24:52 2013 From: mikemol at gmail.com (Michael Mol) Date: Wed, 15 May 2013 12:24:52 -0400 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: References: Message-ID: <5193B6D4.3010402@gmail.com> I sympathize, and I really wish I could, but it's incredibly urgent I get this setup working, as in I need to go from "newbie" to "production" by next week. Since I have it somewhat working as is, I can't afford the time to roll back and try again right now. And since I have very, very little spare resources on the destination end, I can't do a parallel setup...at least not until after I've got everything else working. On 05/15/2013 12:15 PM, Ken Rice wrote: > To figure out whats causing this we really need a ticket opened on jira with > all the good information there... This will allow us to see whats going > on... > > See http://wiki.freeswitch.org/wiki/Reporting_Bugs for more info > > > > On 5/15/13 10:56 AM, "Michael Mol" wrote: > >> I only joined this list today...but I'll note I ran into a problem with >> the YUM-based installation resulting in a freeswitch service which would >> die with SIGABRT (according to strace) when I tried setting it and >> Bluebox up on a CentOS 6.4 machine. Compiling from source does not have >> this issue for me. >> >> >> On 05/15/2013 11:45 AM, Cal Leeming [Simplicity Media Ltd] wrote: >>> Nice timing, Google IO is today :( >>> >>> Cal >>> >>> On Mon, May 13, 2013 at 9:52 PM, Ken Rice >> > wrote: >>> >>> Hey Guys, >>> >>> Just a quick heads up, As you know last week we released FreeSWITCH >>> 1.2.9 and I said packages were coming. >>> >>> We now have both YUM and Debian Repo?s for you guys to install from... >>> >>> For Debian check out >>> >>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_packag >>> es >>> For Centos check out >>> >>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_Ins >>> tallation >>> >>> These will get you the stable branch for now... >>> >>> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and >>> rpm?s for it >>> >>> >>> -- >>> Ken >>> _http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> _irc.freenode.net #freeswitch >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/66d7e9c8/attachment.bin From msc at freeswitch.org Wed May 15 20:44:47 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 09:44:47 -0700 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Message-ID: Hello folks, The weekly conference call will start shortly. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_15 Our VoIPMonitor.org speaker had to reschedule at the last minute so we'll be having a community scrum. I have a few items to discuss as does Ken Rice. If you have some items to discuss please bring those as well. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/329cbaf8/attachment.html From msc at freeswitch.org Wed May 15 20:50:58 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 09:50:58 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: On Tue, May 14, 2013 at 11:44 PM, Mehroz Ashraf wrote: > About FusionPBX, i guess this is not a complete opensource solution, the > dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is good > and support is alive. > Can you expand upon this? I don't recall the dialplan being "hidden" and I'm sure the source code is open for all to see. It may be that the method of handling call routing in the UI is less flexible than some people might need but I don't consider that necessarily a bad thing. A GUI by definition restricts one's options. That's the entire point of a GUI - to abstract away all but the "essential" items. I haven't used Fusion in a while but last time I did it had a place to edit directly the XML files which means you can always make your own custom extensions if need be. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/7b9b4185/attachment.html From krice at freeswitch.org Wed May 15 20:54:30 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 15 May 2013 11:54:30 -0500 Subject: [Freeswitch-users] Ken's News and Notes for FreeSWITCH.... In-Reply-To: <5193B6D4.3010402@gmail.com> Message-ID: Without a stack trace we can not help you. This is the reason we ask for these things... I understand you have a deadline, but I wouldn't even know where you start looking for whats causing your prolbme with the brief description you gave... Its like calling the mechanic and saying my car is making a funny noise then when the mechanic requests you bring it by, you tell him aint nobody got time for that... Unfortunately we get requests like this all the time and theres really nothing we can do... On 5/15/13 11:24 AM, "Michael Mol" wrote: > I sympathize, and I really wish I could, but it's incredibly urgent I > get this setup working, as in I need to go from "newbie" to "production" > by next week. Since I have it somewhat working as is, I can't afford the > time to roll back and try again right now. And since I have very, very > little spare resources on the destination end, I can't do a parallel > setup...at least not until after I've got everything else working. > > On 05/15/2013 12:15 PM, Ken Rice wrote: >> To figure out whats causing this we really need a ticket opened on jira with >> all the good information there... This will allow us to see whats going >> on... >> >> See http://wiki.freeswitch.org/wiki/Reporting_Bugs for more info >> >> >> >> On 5/15/13 10:56 AM, "Michael Mol" wrote: >> >>> I only joined this list today...but I'll note I ran into a problem with >>> the YUM-based installation resulting in a freeswitch service which would >>> die with SIGABRT (according to strace) when I tried setting it and >>> Bluebox up on a CentOS 6.4 machine. Compiling from source does not have >>> this issue for me. >>> >>> >>> On 05/15/2013 11:45 AM, Cal Leeming [Simplicity Media Ltd] wrote: >>>> Nice timing, Google IO is today :( >>>> >>>> Cal >>>> >>>> On Mon, May 13, 2013 at 9:52 PM, Ken Rice >>> > wrote: >>>> >>>> Hey Guys, >>>> >>>> Just a quick heads up, As you know last week we released FreeSWITCH >>>> 1.2.9 and I said packages were coming. >>>> >>>> We now have both YUM and Debian Repo?s for you guys to install from... >>>> >>>> For Debian check out >>>> >>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_pack >>>> ag >>>> es >>>> For Centos check out >>>> >>>> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#YUM_Based_I >>>> ns >>>> tallation >>>> >>>> These will get you the stable branch for now... >>>> >>>> TL;DR Version: FreeSWITCH 1.2.9 is out and so are the deb?s and >>>> rpm?s for it >>>> >>>> >>>> -- >>>> Ken >>>> _http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> _irc.freenode.net #freeswitch >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From msc at freeswitch.org Wed May 15 20:58:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 09:58:02 -0700 Subject: [Freeswitch-users] hangup extension In-Reply-To: References: <5190AC8A.1080903@telecube.com.au> Message-ID: On Tue, May 14, 2013 at 11:28 PM, Ashish gautam wrote: > Hi Michael, > > I set in my dialplan: > > > and in my hangup hook I am doing this: > > sub hupHook{ > freeswitch::consoleLog("err","In Sub hupHook\n"); > my $endDispo = $api->executeString("uuid_getvar $uuid endpoint_disposition > XML"); > chomp($endDispo);# > freeswitch::consoleLog("INFO","Endpoint dispostion huphook is: > $endDispo\n"); > } > > on console its showing this: > > 2013-05-15 11:51:13.157662 [INFO] switch_cpp.cpp:1275 Endpoint dispostion > huphook is: -ERR No such channel! > The channel itself is gone - the endpoints have been disconnected. This means you cannot use uuid_getvar. However, the hangup hook script should receive an object that contains all the information about the channel. According to the wiki you just need to use "our $env;" at the beginning of the script and you should be able to glean everything you need from the $env object. Check this page for details: http://wiki.freeswitch.org/wiki/Mod_perl_Hangup_Hook -MC > > > On Wed, May 15, 2013 at 3:42 AM, Michael Collins wrote: > >> >> >> >> On Mon, May 13, 2013 at 10:31 PM, Ashish gautam wrote: >> >>> Session gets dissolved in hangupHook and we cannot do anything on the >>> basis of UUID >> >> Normally yes, but that's why we have the chan var session_in_hangup_hook. >> Try setting that to true and then using your api_hangup_hook. >> -MC >> >> >>> >>> On Mon, May 13, 2013 at 2:34 PM, Telecube - John wrote: >>> >>>> If you are using Lua script there is ' session:setHangupHook' detailed >>>> here: http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook >>>> >>>> I assume there's something similar in the xml configs >>>> >>>> Regards, >>>> John >>>> >>>> >>>> >>>> On 13/05/13 5:20 PM, Ashish gautam wrote: >>>> >>>> Hi, >>>> >>>> Is there anything like hangup extension which gets executed after the >>>> call is hung up for which the UUID is same and based on that I can do >>>> things for the same call like we have in Asterisk? >>>> >>>> Thanks. >>>> >>>> Regards, >>>> >>>> --Ashish >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/4a66a431/attachment-0001.html From cesar.bermudez at gmail.com Wed May 15 21:05:22 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Wed, 15 May 2013 11:05:22 -0600 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Good info Michael, like usual. Regards. On Wed, May 15, 2013 at 10:50 AM, Michael Collins wrote: > > > > On Tue, May 14, 2013 at 11:44 PM, Mehroz Ashraf > wrote: > >> About FusionPBX, i guess this is not a complete opensource solution, the >> dialplan area is hidden and you cannot change it according to your >> requirements (correct me if i am wrong)... overall functionality is good >> and support is alive. >> > > Can you expand upon this? I don't recall the dialplan being "hidden" and > I'm sure the source code is open for all to see. It may be that the method > of handling call routing in the UI is less flexible than some people might > need but I don't consider that necessarily a bad thing. A GUI by definition > restricts one's options. That's the entire point of a GUI - to abstract > away all but the "essential" items. I haven't used Fusion in a while but > last time I did it had a place to edit directly the XML files which means > you can always make your own custom extensions if need be. > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/343e3c7e/attachment.html From msc at freeswitch.org Wed May 15 21:06:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 10:06:17 -0700 Subject: [Freeswitch-users] channel selection during outbound dialing In-Reply-To: <7CD3228A-FFE5-449A-A6C0-F491EF4C8E58@gmx.net> References: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> <7CD3228A-FFE5-449A-A6C0-F491EF4C8E58@gmx.net> Message-ID: Actually, you have both. Look at your code fragment in wider context: On Wed, May 15, 2013 at 1:42 AM, mbo wrote: > Thanks for the link. It seems that the wiki is not up to date here. In the > code of mod_freetdm I can see that "r" for hunt going downwards starting > from highest channel and "R" hunt going upwards starting from lowest > channel seems to be also available. > > > ... > } else if (*argv[1] == 'r') { > direction = FTDM_RR_DOWN; > } else if (*argv[1] == 'R') { > direction = FTDM_RR_UP; > ? > > Thanks > > Markus > > Am 07.05.2013 um 21:41 schrieb Michael Collins : > > I believe all of that is covered here: > http://wiki.freeswitch.org/wiki/FreeTDM#Dial_Plan > > The "a" means start at channel 1 and work your way up. > -MC > > > On Tue, May 7, 2013 at 10:02 AM, mbo wrote: > >> We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in >> one group. In Asterisk it is possible to specify the channel selection >> mode by a prefix in the dial command (Dial(Zap/g2?) - ascending >> sequential hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2?) - descending >> sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I >> use a dial string like: >> >> freetdm/mygroup/a/ >> >> to dial out via a group. I'm not sure for what the "a" stand for. Are >> there also options for the channel selection mode available like in >> asterisk? >> >> Thanks >> >> Markus >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/be7930fb/attachment.html From msc at freeswitch.org Wed May 15 21:08:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 10:08:20 -0700 Subject: [Freeswitch-users] channel selection during outbound dialing In-Reply-To: References: <4799E98E-DE4E-454F-BE75-1690A886ED21@gmx.net> <7CD3228A-FFE5-449A-A6C0-F491EF4C8E58@gmx.net> Message-ID: Doh, ctrl-enter strikes again. Here we go: Looking at the code fragment in wider context: if (*argv[1] == 'A') { direction = FTDM_BOTTOM_UP; } else if (*argv[1] == 'a') { direction = FTDM_TOP_DOWN; } else if (*argv[1] == 'r') { direction = FTDM_RR_DOWN; } else if (*argv[1] == 'R') { direction = FTDM_RR_UP; You have bottom up, top down, round robin down and round robin up. So yes, if someone could add "r" and "R" to the wiki that would be very helpful. -MC On Wed, May 15, 2013 at 10:06 AM, Michael Collins wrote: > Actually, you have both. Look at your code fragment in wider context: > > > > > On Wed, May 15, 2013 at 1:42 AM, mbo wrote: > >> Thanks for the link. It seems that the wiki is not up to date here. In >> the code of mod_freetdm I can see that "r" for hunt going downwards >> starting from highest channel and "R" hunt going upwards starting from >> lowest channel seems to be also available. >> >> >> ... >> } else if (*argv[1] == 'r') { >> direction = FTDM_RR_DOWN; >> } else if (*argv[1] == 'R') { >> direction = FTDM_RR_UP; >> ? >> >> Thanks >> >> Markus >> >> Am 07.05.2013 um 21:41 schrieb Michael Collins : >> >> I believe all of that is covered here: >> http://wiki.freeswitch.org/wiki/FreeTDM#Dial_Plan >> >> The "a" means start at channel 1 and work your way up. >> -MC >> >> >> On Tue, May 7, 2013 at 10:02 AM, mbo wrote: >> >>> We have a sangoma A104d QUAD T1/E1 AFT card configured with 4 spans in >>> one group. In Asterisk it is possible to specify the channel selection >>> mode by a prefix in the dial command (Dial(Zap/g2?) - ascending >>> sequential hunt group Looks in order 1, 2, 5, 8), Dial(Zap/G2?) - descending >>> sequential hunt group, looks in order 8, 5, 2, 1), etc. In freeswitch I >>> use a dial string like: >>> >>> freetdm/mygroup/a/ >>> >>> to dial out via a group. I'm not sure for what the "a" stand for. Are >>> there also options for the channel selection mode available like in >>> asterisk? >>> >>> Thanks >>> >>> Markus >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/48ad91aa/attachment-0001.html From anthony.minessale at gmail.com Wed May 15 21:23:15 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 15 May 2013 12:23:15 -0500 Subject: [Freeswitch-users] JIRA contributions - 15/05/2013 In-Reply-To: References: Message-ID: I like the needs info for sure, I've wanted to go as far as heuristic tests to auto detect that it needs info ;) The auto resolve from that state is a good one too. We should for sure make better categories than we have for bounty or things that are cool but there is no time to do. Right now we have a user who's name is "make me stuff for free" and we assign them to him so that can be improved. The 5 for 1 thing is good, It would be great to get everyone just doing it naturally where they can learn to scan a jira in a once-over and be able to tell what else it needs. I'm also interested in maybe some way to get the policies and other info visible to the users so we don't have to keep repeating so many things. On Wed, May 15, 2013 at 11:07 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > Last week I made some contributions on JIRA by triaging some tickets. > > FS-3964 - requested larger bounty offer, if no bite then reclassify as > nice to have > FS-4211 - requested further info, close after 2 weeks if no response. > FS-4281 - requested more info, waiting on core devs > FS-4891 - requested more info, updated wiki docs, close after 2 weeks if > no response > FS-5154 - requested OP create new ticket for seperate issues, close after > 2 weeks if no response > > However, I was unable to action them due to lack of access on JIRA. I sent > an email to Ken/Michael requesting access to close tickets and mark as > resolved etc, but realised this was not the appropriate way as you > shouldn't need to have extra privileges in order to do ticket triage. > > Therefore, I'd like to open up discussion about how we can improve JIRA to > allow for public contributions and make ticket triaging easier. > > My suggestions are; > > * Add a new step in the workflow that switches it into "needs more client > info", so it automatically closes the ticket after X days if no extra info > is received. > * Have a "maybe one day/needs contributions" status in the workflow, > rather than leaving the ticket as open.. that way we can see really how > many are waiting, and how many are waiting on contributions. > * Have a "needs design decision" status in the workflow, indicating that a > core dev needs to make the final call > * Introduce an official 5-for-1 ticket system where users can triage 5 > tickets and request that a core dev look at any ticket of their choice in > return. > > I think we stand a much better chance of getting a lot more public > contributions if we implement these changes. > > Probably a bit too short notice to discuss on todays conf call, but it > would be good to get some ideas/feedback in time for next weeks call. > Modifying JIRA and making workflow changes is no easy task, so it would > need to be thoroughly discussed and understood. I'll be on the call today > 20 minutes early today if anyone wants to discuss. > > Any thoughts/ideas? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/040299bb/attachment.html From jmesquita at freeswitch.org Wed May 15 21:23:57 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 15 May 2013 14:23:57 -0300 Subject: [Freeswitch-users] JIRA contributions - 15/05/2013 In-Reply-To: References: Message-ID: Your first suggestion is already available. That's what I do every single day and most of the times, I get no response and tickets get closed automatically. I guess that more than doing triage we lack the human power to reproduce on a controlled environment the tickets that get unattended... Just my 2 cents Jo?o Mesquita FreeSWITCH? Solutions On Wed, May 15, 2013 at 1:07 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > Last week I made some contributions on JIRA by triaging some tickets. > > FS-3964 - requested larger bounty offer, if no bite then reclassify as > nice to have > FS-4211 - requested further info, close after 2 weeks if no response. > FS-4281 - requested more info, waiting on core devs > FS-4891 - requested more info, updated wiki docs, close after 2 weeks if > no response > FS-5154 - requested OP create new ticket for seperate issues, close after > 2 weeks if no response > > However, I was unable to action them due to lack of access on JIRA. I sent > an email to Ken/Michael requesting access to close tickets and mark as > resolved etc, but realised this was not the appropriate way as you > shouldn't need to have extra privileges in order to do ticket triage. > > Therefore, I'd like to open up discussion about how we can improve JIRA to > allow for public contributions and make ticket triaging easier. > > My suggestions are; > > * Add a new step in the workflow that switches it into "needs more client > info", so it automatically closes the ticket after X days if no extra info > is received. > * Have a "maybe one day/needs contributions" status in the workflow, > rather than leaving the ticket as open.. that way we can see really how > many are waiting, and how many are waiting on contributions. > * Have a "needs design decision" status in the workflow, indicating that a > core dev needs to make the final call > * Introduce an official 5-for-1 ticket system where users can triage 5 > tickets and request that a core dev look at any ticket of their choice in > return. > > I think we stand a much better chance of getting a lot more public > contributions if we implement these changes. > > Probably a bit too short notice to discuss on todays conf call, but it > would be good to get some ideas/feedback in time for next weeks call. > Modifying JIRA and making workflow changes is no easy task, so it would > need to be thoroughly discussed and understood. I'll be on the call today > 20 minutes early today if anyone wants to discuss. > > Any thoughts/ideas? > > Thanks > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/7056d850/attachment.html From mikemol at gmail.com Wed May 15 22:07:17 2013 From: mikemol at gmail.com (Michael Mol) Date: Wed, 15 May 2013 14:07:17 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <5193B6B6.4080907@freeswitch.org> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> <5193B6B6.4080907@freeswitch.org> Message-ID: <5193CED5.5060306@gmail.com> Perhaps what's needed are DNSBL services, similar to what's used with email? (The difference being, if I were using a DNSBL, I'd pass flagged calls directly into voicemail, perhaps even a secondary voicemail box, rather than allowing it to actively interrupt anyone.) On 05/15/2013 12:24 PM, I put the Who? in Mishehu wrote: > There are already sites like this up, but they don't act as a > blacklist. More so they're forums where people talk about what type of > calls they received from a given phone number. > > -Yossi > > On 05/15/2013 08:11 AM, Telecube - John wrote: >> I can imagine something like that getting a bit messy, what if some >> smarty pants decides to poison it by randomly generating numbers and >> sending them to your list? >> >> - John >> >> On 15/05/13 10:42 PM, Michael Jerris wrote: >>> Anyone interested in us setting up a web service that you can post >>> naughty numbers too so we can share? This could be a fun project to >>> set up. >>> >>> On May 15, 2013, at 7:03 AM, Andrew Cassidy >>> >> > wrote: >>> >>>> Indeed. I work for one of the companies that buys personal data then >>>> sells it on again... >>>> >>>> >>>> On 15 May 2013 11:35, Steven Ayre >>> > wrote: >>>> >>>> Yep. Unfortunately as already noted its not perfect - plenty of >>>> people ignore it (afaik illegally, call from outside the UK, or >>>> most inciduously have permission to call you because they bought >>>> the number from someone who bought the number from someone >>>> who bought the number from someone who you accidentally ticked >>>> the 'you and anyone you sell my number to can call me' tick >>>> boxes on a form 15 years ago. >>>> >>>> -Steve >>>> >>>> >>>> On Wednesday, May 15, 2013, Andrew Cassidy wrote: >>>> >>>> With regards to the first question, one of my old university >>>> lecturers plays a not in service tone. >>>> >>>> We have something similar in the >>>> uk: http://www.tpsonline.org.uk/tps/index.html >>>> >>>> >>>> On 15 May 2013 03:21, Michael Collins >>>> wrote: >>>> >>>> Are you using the example configs? If so you can make a >>>> local call, like to 9664, and then run hash_dump again. >>>> -MC >>>> >>>> >>>> On Tue, May 14, 2013 at 6:50 PM, Sean Devoy >>>> wrote: >>>> >>>> Thanks MC I will look at the call return in the >>>> examples. >>>> >>>> >>>> >>>> But: hash_dump all >>>> >>>> -ERR no reply >>>> >>>> >>>> >>>> *From:*freeswitch-users-bounces at lists.freeswitch.org >>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] >>>> *On Behalf Of *Michael Collins >>>> *Sent:* Tuesday, May 14, 2013 7:53 PM >>>> *To:* FreeSWITCH Users Help >>>> *Subject:* Re: [Freeswitch-users] Late mother's day >>>> gift - Advanced Do Not Call service! >>>> >>>> >>>> >>>> In the example config there's already a "call >>>> return" feature that utilizes the last received >>>> caller ID value. If there was a valid caller ID >>>> number sent then it will be stored in the local >>>> database, which is where the call_return extension >>>> gets it from. >>>> >>>> -MC >>>> >>>> P.S. - If you haven't already started using the FS >>>> local database then check it out. Try the "hash_dump >>>> all" command at fs_cli to get an idea of what gets >>>> stored. >>>> >>>> >>>> >>>> On Tue, May 14, 2013 at 3:08 PM, Sean Devoy >>>> wrote: >>>> >>>> >>>> >>>> HI all, >>>> >>>> >>>> >>>> My mother has made a request and Mom?s always get >>>> our best effort right? >>>> >>>> >>>> >>>> For those of you outside the US, we have a national >>>> Do Not Call Registry: www.donotcall.gov >>>> . You register your >>>> number there and in 30 days or less you are on the >>>> list. Telemarketers are required by law to check >>>> the list before calling and not call if you are on it. >>>> >>>> >>>> >>>> It does certainly help, but overseas call centers >>>> tend to ignore it and of course the excluded groups >>>> from the law ?*charities, political organizations, >>>> and telephone surveyors*? may still call you. >>>> >>>> >>>> >>>> So it is simple enough for her to tell >>>> >>>> *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 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>>> >>>> >>>> -- >>>> *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 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/d992b413/attachment-0001.bin From mikemol at gmail.com Wed May 15 22:12:44 2013 From: mikemol at gmail.com (Michael Mol) Date: Wed, 15 May 2013 14:12:44 -0400 Subject: [Freeswitch-users] RPMs and SIGABRT In-Reply-To: References: Message-ID: <5193D01C.9090804@gmail.com> On 05/15/2013 12:54 PM, Ken Rice wrote: > Without a stack trace we can not help you. This is the reason we ask for > these things... > > I understand you have a deadline, but I wouldn't even know where you start > looking for whats causing your prolbme with the brief description you > gave... Its like calling the mechanic and saying my car is making a funny > noise then when the mechanic requests you bring it by, you tell him aint > nobody got time for that... > > Unfortunately we get requests like this all the time and theres really > nothing we can do... Understand I'm not asking for assistance resolving that particular issue. I found my own workaround by compiling from git HEAD instead. I only posted that report as a sort of tally mark in case someone else had the problem, as compiling from git HEAD worked as an alternate solution. Have no fear, I intend to return with a proper report, if time permits. (That said, I've got at least half a dozen bug reports open on bugs.gentoo.org based with similar intentions...) [snip; apologies, but writing top posts just feels wrong, even when continuing a thread's trend.] -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/7218af75/attachment.bin From acrow at integrafin.co.uk Wed May 15 23:03:41 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 15 May 2013 20:03:41 +0100 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: <5193DC0D.9000406@integrafin.co.uk> Hi Mehroz, all, Having used FusionPBX, I can tell you it's not closed source in any place. It's just a load of PHP code and none of it is obfuscated or encrypted. It actually contains an XML editor component so you can add and edit XML dialplans, configurations, in fact anything under your /etc/freeswitch directory without limits. Just make sure they don't clash with the database-generated ones (ie the ones you create in the simple "dialplan GUI", start with v_ as I remember) and you will be fine. It's very flexible IMHO, but if you want to stop users getting at those parts they don't need, either put them in the appropriate readymade access group, create your own, or indeed change the code. Personally if I was using it now, I'd create the more "icky" stuff in the XML editor and then let the simple stuff (ie extensions, callgroups, etc) for the GUI. It was my intro to FreeSWITCH and actually a good one, as it doesn't hide too much, teaches you the logic of dialplans, and better than that you can see the XML result of what you did in the "ID10T" part of it just by moving to a different tab. It's also well structured enough that if something goes wrong and you know anything (even basic) about programming you can figure out where it happened and fix it yourself or at least give a good bug report. Cheers Alex On 15/05/13 18:05, Cesar Bermudez wrote: > Good info Michael, like usual. > Regards. > > On Wed, May 15, 2013 at 10:50 AM, Michael Collins > wrote: > > > > > On Tue, May 14, 2013 at 11:44 PM, Mehroz Ashraf > > wrote: > > About FusionPBX, i guess this is not a complete opensource > solution, the dialplan area is hidden and you cannot change it > according to your requirements (correct me if i am wrong)... > overall functionality is good and support is alive. > > > Can you expand upon this? I don't recall the dialplan being > "hidden" and I'm sure the source code is open for all to see. It > may be that the method of handling call routing in the UI is less > flexible than some people might need but I don't consider that > necessarily a bad thing. A GUI by definition restricts one's > options. That's the entire point of a GUI - to abstract away all > but the "essential" items. I haven't used Fusion in a while but > last time I did it had a place to edit directly the XML files > which means you can always make your own custom extensions if need be. > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/bb7f9899/attachment.html From d at d-man.org Wed May 15 23:30:56 2013 From: d at d-man.org (Darren Schreiber) Date: Wed, 15 May 2013 12:30:56 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: Message-ID: Hi Mehroz, There is an issue with the current git and nobody here has had the time to fix it, but the ISO works. That said, the project has become dated, I agree. It's definitely not dead though. We are actually rebuilding both blue.box items this week (the ISO and fixing git), which should fix the build. It's definitely not dead. As of last week, Francis has been dedicated full-time on blue.box. You will see activity on it via that route. You can also always ask questions in #2600hz. - Darren From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Tuesday, May 14, 2013 11:44 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/a0d903db/attachment.html From acrow at integrafin.co.uk Wed May 15 23:35:34 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 15 May 2013 20:35:34 +0100 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <5193B6B6.4080907@freeswitch.org> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> <5193B6B6.4080907@freeswitch.org> Message-ID: <5193E386.9070503@integrafin.co.uk> The idea of a CLI reputation service comes to mind. You'd have to have something to make sure that submissions tp the service are sufficiently difficult for voice spammers but easy for legitimate originators. However such a thing requires traction to work (just like DNSSEC, SPF and DomainKeys) so it might be a long wait... On 15/05/13 17:24, I put the Who? in Mishehu wrote: > There are already sites like this up, but they don't act as a > blacklist. More so they're forums where people talk about what type > of calls they received from a given phone number. > > -Yossi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/e19bd417/attachment-0001.html From msc at freeswitch.org Thu May 16 00:18:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 13:18:02 -0700 Subject: [Freeswitch-users] Decline In-Reply-To: References: Message-ID: I threw this up on our pastebin because these old eye need colorized output. :) http://pastebin.freeswitch.org/20922 I only see SIP messages between FS and the phone. Was there no SIP traffic between FS and the carrier? Or did you perhaps have SIP trace on the internal profile but not the external profile? If possible recreate the issue with siptrace on both internal and external profiles. Thanks! -MC On Wed, May 15, 2013 at 8:48 AM, Blake Priddy wrote: > Ok I talked to you guys previously about this and our gateway told me I am > sending the call to them incorrectly.. Which it is just 2 phones that are > doing this. The rest are fine? > > http://pastebin.com/3ppLaatg > > Any thing in there throw a flag? > > -- > > *Blakelund Priddy* > Network & Systems Engineer > Bryant Public School District > Bryant, Arkansas 72022 > http://www.bryantschools.org > p 501-653-5038 > f 501-847-5656 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/19e36ee2/attachment.html From msc at freeswitch.org Thu May 16 00:19:49 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 13:19:49 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Darren, Thanks for the heads up! We will be on the lookout for more blue.box stuff. -MC On Wed, May 15, 2013 at 12:30 PM, Darren Schreiber wrote: > Hi Mehroz, > There is an issue with the current git and nobody here has had the time to > fix it, but the ISO works. > > That said, the project has become dated, I agree. It's definitely not dead > though. We are actually rebuilding both blue.box items this week (the ISO > and fixing git), which should fix the build. It's definitely not dead. > > As of last week, Francis has been dedicated full-time on blue.box. You > will see activity on it via that route. > > You can also always ask questions in #2600hz. > > - Darren > > From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Date: Tuesday, May 14, 2013 11:44 PM > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX > > About FusionPBX, i guess this is not a complete opensource solution, the > dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is good > and support is alive. > > and yes, bluebox seems to be dead now, howeverm very nice interface, > completely opensource. i am having some issues in its customization and > there seems to be no support alive! > > also note that, the iso image has different bluebox interface and git > version has different! and no one to help! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/f1adf18e/attachment.html From lndspereira-fs at yahoo.com Thu May 16 00:56:54 2013 From: lndspereira-fs at yahoo.com (Leonardo Pereira) Date: Wed, 15 May 2013 13:56:54 -0700 (PDT) Subject: [Freeswitch-users] UUID in MESSAGE event Message-ID: <1368651414.81263.YahooMailNeo@web125803.mail.ne1.yahoo.com> Hi. I'm using the 'chat' command to send SMS: ??? bgapi chat sip|+13229944553|external/+551991221234 at MY.GW.COM|test message Is there a way to have some kind of UUID parameter in MESSAGE event? ? Leonardo Nogueira de S? Pereira Tel.: +55 19 3307-5589 Cel.: +55 19 9122-5943 Skype: leonardo_pereira_77 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/90f03293/attachment.html From sdevoy at bizfocused.com Thu May 16 01:00:06 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 15 May 2013 17:00:06 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <51938969.5070707@telecube.com.au> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> Message-ID: <071401ce51af$2dc847b0$8958d710$@bizfocused.com> Thanks All. Still wondering about why my FS does not like hash_dump all and returns "-ERR no reply". Perhaps when I rebuild on the current/stable branch I will have better luck. I LOVE the transfer to fax_recv option. John - I like the way you think. I could setup a web service that: . worked like a moderated list and give out ids to just our "friends". . Had an "add" url that required your id/password to add a banned number . Had a query url to return "TRUE" is the number is in the list, else FALSE Alternatively, we could track who has reported the number. If 4 or more report the number, add it to the "public" list, else just return it as banned to the people who reported it. Any interest? We could set about defining the interface. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Telecube - John Sent: Wednesday, May 15, 2013 9:11 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! I can imagine something like that getting a bit messy, what if some smarty pants decides to poison it by randomly generating numbers and sending them to your list? - John On 15/05/13 10:42 PM, Michael Jerris wrote: Anyone interested in us setting up a web service that you can post naughty numbers too so we can share? This could be a fun project to set up. On May 15, 2013, at 7:03 AM, Andrew Cassidy wrote: Indeed. I work for one of the companies that buys personal data then sells it on again... On 15 May 2013 11:35, Steven Ayre wrote: Yep. Unfortunately as already noted its not perfect - plenty of people ignore it (afaik illegally, call from outside the UK, or most inciduously have permission to call you because they bought the number from someone who bought the number from someone who bought the number from someone who you accidentally ticked the 'you and anyone you sell my number to can call me' tick boxes on a form 15 years ago. -Steve On Wednesday, May 15, 2013, Andrew Cassidy wrote: With regards to the first question, one of my old university lecturers plays a not in service tone. We have something similar in the uk: http://www.tpsonline.org.uk/tps/index.html On 15 May 2013 03:21, Michael Collins wrote: Are you using the example configs? If so you can make a local call, like to 9664, and then run hash_dump again. -MC On Tue, May 14, 2013 at 6:50 PM, Sean Devoy wrote: Thanks MC I will look at the call return in the examples. But: hash_dump all -ERR no reply From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, May 14, 2013 7:53 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In the example config there's already a "call return" feature that utilizes the last received caller ID value. If there was a valid caller ID number sent then it will be stored in the local database, which is where the call_return extension gets it from. -MC P.S. - If you haven't already started using the FS local database then check it out. Try the "hash_dump all" command at fs_cli to get an idea of what gets stored. On Tue, May 14, 2013 at 3:08 PM, Sean Devoy wrote: HI all, My mother has made a request and Mom's always get our best effort right? For those of you outside the US, we have a national Do Not Call Registry: www.donotcall.gov . You register your number there and in 30 days or less you are on the list. Telemarketers are required by law to check the list before calling and not call if you are on it. It does certainly help, but overseas call centers tend to ignore it and of course the excluded groups from the law "charities, political organizations, and telephone surveyors" may still call you. So it is simple enough for her to tell 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130515/69368f80/attachment-0001.html From tru083 at yahoo.com Wed May 15 23:26:18 2013 From: tru083 at yahoo.com (D D) Date: Wed, 15 May 2013 12:26:18 -0700 (PDT) Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> Message-ID: <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> Hi, I am using H264 codec for a call into FS, and recording it to an FSV file. How can I convert the FSV file to MP4? ? Any ideas on how to get started on this task would be greatly appreciated! Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/c9bec969/attachment.html From ehermouet at bluetel.fr Thu May 16 01:01:17 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Wed, 15 May 2013 23:01:17 +0200 Subject: [Freeswitch-users] DTMF outbound call Message-ID: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> Hi all. We have voip provider connectd to FS and 1 extension. When my extension make call using voip provider dtmf not work. Hi phone to voip provider and dtmf is active to rfc2833. So i suppose i miss something on my config. Can you explain where ? tks advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/6fbd03cf/attachment.html From msc at freeswitch.org Thu May 16 01:39:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 14:39:17 -0700 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: <071401ce51af$2dc847b0$8958d710$@bizfocused.com> References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> <071401ce51af$2dc847b0$8958d710$@bizfocused.com> Message-ID: On Wed, May 15, 2013 at 2:00 PM, Sean Devoy wrote: > Thanks All.**** > > ** ** > > Still wondering about why my FS does not like hash_dump all and returns ?-ERR > no reply?. Perhaps when I rebuild on the current/stable branch I will have > better luck. > I suspect that you may have modified the example configs to the point where they are not putting anything into the local database. In any case you can test some of the commands right from fs_cli. See this wiki pagefor details, but here's a simple insert command: hash insert/my_realm/foo/bar Try that and then do hash_dump and see what shakes out. -MC > **** > > ** ** > > I LOVE the transfer to fax_recv option. John ? I like the way you think.* > *** > > ** ** > > I could setup a web service that:**** > > **? **worked like a moderated list and give out ids to just our > ?friends?.**** > > **? **Had an ?add? url that required your id/password to add a > banned number**** > > **? **Had a query url to return ?TRUE? is the number is in the > list, else FALSE**** > > ** ** > > Alternatively, we could track who has reported the number. If 4 or more > report the number, add it to the ?public? list, else just return it as > banned to the people who reported it. **** > > ** ** > > Any interest? We could set about defining the interface.**** > > ** ** > > Sean**** > > ** ** > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/b76c627f/attachment.html From msc at freeswitch.org Thu May 16 01:42:32 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 14:42:32 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> Message-ID: I recommend that you get a console debug log of the call and put it up on pastebin.freeswitch.org. Hopefully we'll be able to see what is happening. In the meantime, is your provider set to use RFC2833 digits or do they want to see in-band DTMFs? -MC On Wed, May 15, 2013 at 2:01 PM, Erwan Hermouet wrote: > Hi all. We have voip provider connectd to FS and 1 extension. When my > extension make call using voip provider dtmf not work. Hi phone to voip > provider and dtmf is active to rfc2833.**** > > ** ** > > So i suppose i miss something on my config. Can you explain where ? tks > advance**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/ce5c3f9e/attachment.html From sdevoy at bizfocused.com Thu May 16 01:51:53 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 15 May 2013 17:51:53 -0400 Subject: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! In-Reply-To: References: <290d01ce50ef$8d10b010$a7321030$@bizfocused.com> <00f101ce510e$8fab18f0$af014ad0$@bizfocused.com> <51938969.5070707@telecube.com.au> <071401ce51af$2dc847b0$8958d710$@bizfocused.com> Message-ID: <079401ce51b6$698fe9e0$3cafbda0$@bizfocused.com> Thanks MC. It is all completely clear now. Sorry for being slow. Now to revisit the examples now that I may understand what is happening! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 15, 2013 5:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Late mother's day gift - Advanced Do Not Call service! On Wed, May 15, 2013 at 2:00 PM, Sean Devoy wrote: Thanks All. Still wondering about why my FS does not like hash_dump all and returns "-ERR no reply". Perhaps when I rebuild on the current/stable branch I will have better luck. I suspect that you may have modified the example configs to the point where they are not putting anything into the local database. In any case you can test some of the commands right from fs_cli. See this wiki page for details, but here's a simple insert command: hash insert/my_realm/foo/bar Try that and then do hash_dump and see what shakes out. -MC I LOVE the transfer to fax_recv option. John - I like the way you think. I could setup a web service that: . worked like a moderated list and give out ids to just our "friends". . Had an "add" url that required your id/password to add a banned number . Had a query url to return "TRUE" is the number is in the list, else FALSE Alternatively, we could track who has reported the number. If 4 or more report the number, add it to the "public" list, else just return it as banned to the people who reported it. Any interest? We could set about defining the interface. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/0883dc44/attachment-0001.html From mishehu at freeswitch.org Thu May 16 02:28:40 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Wed, 15 May 2013 17:28:40 -0500 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> Message-ID: <51940C18.10603@freeswitch.org> For multimedia format conversion needs, I generally recommend ffmpeg. Please see http://www.ffmpeg.org/ It's a little more complex but you can script it very easily. Otherwise if you use KDE you can try out http://www.kdenlive.org/ for editing and transcoding. -Yossi On 05/15/2013 02:26 PM, D D wrote: > Hi, > > I am using H264 codec for a call into FS, and recording it to an FSV file. > > How can I convert the FSV file to MP4? > > Any ideas on how to get started on this task would be greatly appreciated! > > Thanks, > David > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/52ce4f0c/attachment.html From nick at hkcradio.com Thu May 16 04:23:43 2013 From: nick at hkcradio.com (Nick Giannak) Date: Wed, 15 May 2013 20:23:43 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: <5194270F.20407@hkcradio.com> Darren, Can you do us a favor and post here when it's fixed, so we can play with it again? You once recommended Kazoo to me but it was so far beyond the scope of what I needed or knew what to do with. It's nice to see blue.box back in development! Thanks, Nick On 5/15/2013 3:30 PM, Darren Schreiber wrote: > Hi Mehroz, > There is an issue with the current git and nobody here has had the > time to fix it, but the ISO works. > > That said, the project has become dated, I agree. It's definitely not > dead though. We are actually rebuilding both blue.box items this week > (the ISO and fixing git), which should fix the build. It's definitely > not dead. > > As of last week, Francis has been dedicated full-time on blue.box. You > will see activity on it via that route. > > You can also always ask questions in #2600hz. > > - Darren > > From: Mehroz Ashraf > > Reply-To: "freeswitch-users at lists.freeswitch.org > " > > > Date: Tuesday, May 14, 2013 11:44 PM > To: "freeswitch-users at lists.freeswitch.org > " > > > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX > > About FusionPBX, i guess this is not a complete opensource solution, > the dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is > good and support is alive. > > and yes, bluebox seems to be dead now, howeverm very nice interface, > completely opensource. i am having some issues in its customization > and there seems to be no support alive! > > also note that, the iso image has different bluebox interface and git > version has different! and no one to help! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130515/35e5f1eb/attachment.html From sirimmfs at gmail.com Thu May 16 04:33:54 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 16 May 2013 10:33:54 +1000 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: <23490192.5864.1368605939908.JavaMail.javamailuser@localhost> References: <15534432.5814.1368605789796.JavaMail.javamailuser@localhost> <23490192.5864.1368605939908.JavaMail.javamailuser@localhost> Message-ID: Hey Ray, Thanks for the response. I agree that this is a strange scenario - there seems to be some problem with the DHCP server in the environmnet, because of which, after lease time expiry, it sometimes doesn't provide back the IP to the device. I assume this triggers a interface down scenario in the device, causing the " loose ethX and FS to all available interfaces" scenario. I have a couple of questions though: 1. " loose ethX and you are binding FS to all available interfaces" - Is FS binding to available interfaces part configurable? Can I ask freeswitch not to bind to loop back interface if eth0 goes down? 2. Keeping in mind that this is a strange scenario, should FS still behave this way? I mean, once the device has got back the right IP, I wonder why the internal profile has not been loaded back? Thanks again mate! On Wed, May 15, 2013 at 6:19 PM, info wrote: > On Tue, May 14, 2013 at 10:31 AM, Siri MM wrote: > >> Hello All, >> >> I am facing an issue with freeswitch installed on my device. For some >> reason, IP address on my device sometimes seems to be 'fluctuating' between >> the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this >> change everytime, and reconfigure accordingly. However, during one such >> occassion, it hasn't brought up the internal profile, and isn't binding to >> port 5060. No other program uses this port on my device. >> >> Working log: >> http://pastebin.freeswitch.org/20912 >> >> Not-working log: >> http://pastebin.freeswitch.org/20915 >> >> Would appreciate any inputs >> >> Thanks! >> > > Hi Siri, > > I am a bit mystified as to how things are supposed to work if (how?) your > IP get's switched between static and 127.0.0.1 (again, how?) This feels > just wrong on so many levels :-) > > 127.0.0.1 normally is bound to the loopback interface, which, on linux, is > called lo > your other IP should be bound to an eth0 interface, or an internal ip and > and the static IP on a router with some sort of NAT or port redirection. > AFAIK linux should not allow you bind 127.0.0.1 to eth0 if it is allready > bound to lo, so my best guess would be that you loose ethX and you are > binding FS to all available interfaces... > > Alas, I don't know how to answer, because even if there should be a > problem bringing up the internal profile on such a "switch", my guess is > that that's probably *not* the problem you should be looking to fix for > right now. > > Especially since, if somehow the IP really get's switched to 127.0.0.1, > you will be talking only to yourself and I do not see how this should > create a working setup (unless you use only an internal sip/(pri|bri) card > or some USB channelbanks/gprs sticks, etc. and you only will be doing > robocalls or automated IVR's ... > > Well, sorry mate that I can't help you more, but this really is a strange > szenario, > > best regards, > Ray > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/e6d0786c/attachment-0001.html From sirimmfs at gmail.com Thu May 16 04:35:55 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 16 May 2013 10:35:55 +1000 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: References: Message-ID: Hi Karsten, Thanks for the reply. I am working in a DHCP environment, hence I assume I will no tbe able to disable auto-restart, as the IP could change. Please let me know your view. Thanks! On Wed, May 15, 2013 at 6:27 PM, Karsten Horsmann wrote: > Hi Siri, > > try to disable auto-restart in mod_sofia profile and fill in the internal > ip-address. > > > > > > 2013/5/15 Siri MM > >> Hey Guys, Would appreciate any inputs! Thanks. >> >> >> On Tue, May 14, 2013 at 10:31 AM, Siri MM wrote: >> >>> Hello All, >>> >>> I am facing an issue with freeswitch installed on my device. For some >>> reason, IP address on my device sometimes seems to be 'fluctuating' between >>> the actual Static IP, and 127.0.0.1. Freeswitch is able to detect this >>> change everytime, and reconfigure accordingly. However, during one such >>> occassion, it hasn't brought up the internal profile, and isn't binding to >>> port 5060. No other program uses this port on my device. >>> >>> Working log: >>> http://pastebin.freeswitch.org/20912 >>> >>> Not-working log: >>> http://pastebin.freeswitch.org/20915 >>> >>> Would appreciate any inputs >>> >>> Thanks! >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Mit freundlichen Gr??en > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/64fc1611/attachment.html From cal.leeming at simplicitymedialtd.co.uk Thu May 16 04:56:15 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Thu, 16 May 2013 01:56:15 +0100 Subject: [Freeswitch-users] JIRA contributions - 15/05/2013 In-Reply-To: References: Message-ID: Hi guys, Thanks for taking the time to discuss this on the call today. I think the general outcome can be summarized as; * The 5-for-1 approach would need internal discussion before being officially adopted, however some core devs would be willing to participate prior to that happening, on a case-by-case basis * JIRA needs to be improved, and internal discussions will take place to find way for this to be done. * If you cannot triage a ticket due to lack of access, add your comments with the suggested action so a core dev only had to spend a few seconds on the ticket, rather than a few minutes in triage. * Keep calm and carry on, all contributions no matter how small are very much welcomed and appreciated If anyone else has any suggestions on how things can be improved, put your ideas into this thread whilst it's still hot! In the mean time, I have added an unofficial '5-for-1' section into the mailing list with some general guidelines on how to request this, and made it clear that this is not an officially adopted approach yet so YMMV etc. http://wiki.freeswitch.org/wiki/UsingTheMailingList If there is anyone here that wants to contribute but doesn't know where to start, speak up now! Tell us what you want to know, and what we can do to help you contribute. Thanks guys Cal On Wed, May 15, 2013 at 5:07 PM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > Last week I made some contributions on JIRA by triaging some tickets. > > FS-3964 - requested larger bounty offer, if no bite then reclassify as > nice to have > FS-4211 - requested further info, close after 2 weeks if no response. > FS-4281 - requested more info, waiting on core devs > FS-4891 - requested more info, updated wiki docs, close after 2 weeks if > no response > FS-5154 - requested OP create new ticket for seperate issues, close after > 2 weeks if no response > > However, I was unable to action them due to lack of access on JIRA. I sent > an email to Ken/Michael requesting access to close tickets and mark as > resolved etc, but realised this was not the appropriate way as you > shouldn't need to have extra privileges in order to do ticket triage. > > Therefore, I'd like to open up discussion about how we can improve JIRA to > allow for public contributions and make ticket triaging easier. > > My suggestions are; > > * Add a new step in the workflow that switches it into "needs more client > info", so it automatically closes the ticket after X days if no extra info > is received. > * Have a "maybe one day/needs contributions" status in the workflow, > rather than leaving the ticket as open.. that way we can see really how > many are waiting, and how many are waiting on contributions. > * Have a "needs design decision" status in the workflow, indicating that a > core dev needs to make the final call > * Introduce an official 5-for-1 ticket system where users can triage 5 > tickets and request that a core dev look at any ticket of their choice in > return. > > I think we stand a much better chance of getting a lot more public > contributions if we implement these changes. > > Probably a bit too short notice to discuss on todays conf call, but it > would be good to get some ideas/feedback in time for next weeks call. > Modifying JIRA and making workflow changes is no easy task, so it would > need to be thoroughly discussed and understood. I'll be on the call today > 20 minutes early today if anyone wants to discuss. > > Any thoughts/ideas? > > Thanks > > Cal > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/e8660dd5/attachment.html From msc at freeswitch.org Thu May 16 09:08:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 May 2013 22:08:39 -0700 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: References: <15534432.5814.1368605789796.JavaMail.javamailuser@localhost> <23490192.5864.1368605939908.JavaMail.javamailuser@localhost> Message-ID: On Wed, May 15, 2013 at 5:33 PM, Siri MM wrote: > Hey Ray, > > Thanks for the response. I agree that this is a strange scenario - there > seems to be some problem with the DHCP server in the environmnet, because > of which, after lease time expiry, it sometimes doesn't provide back the IP > to the device. I assume this triggers a interface down scenario in the > device, causing the " loose ethX and FS to all available interfaces" > scenario. I have a couple of questions though: > 1. " loose ethX and you are binding FS to all available interfaces" - Is > FS binding to available interfaces part configurable? Can I ask freeswitch > not to bind to loop back interface if eth0 goes down? > I believe another poster mentioned setting the SIP profile param 'auto-restart' to false. Try that first. > 2. Keeping in mind that this is a strange scenario, should FS still behave > this way? I mean, once the device has got back the right IP, I wonder why > the internal profile has not been loaded back? > This is indeed interesting. Does your ethX port actually get 127.0.0.1? Or does Sofia just somehow end up on the lo interface? In any event, if you disable the auto-restart feature it will prevent FS from bouncing over to 127.0.0.1. -Michael -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130515/e3c61cdd/attachment.html From dujinfang at gmail.com Thu May 16 09:43:56 2013 From: dujinfang at gmail.com (Seven Du) Date: Thu, 16 May 2013 13:43:56 +0800 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: <51940C18.10603@freeswitch.org> References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> <51940C18.10603@freeswitch.org> Message-ID: fsv is not supported in standard ffmpeg, someone did some work on this and possibly in the freeswitch_contrib repo, also you may find it by following this: https://www.google.com.hk/search?safe=strict&rlz=1C5CHFA_enCN506CN506&q=lists.freeswitch.org+fsvdec.c&oq=lists.freeswitch.org+fsvdec.c&gs_l=serp.3...8140.13246.0.13663.21.21.0.0.0.3.153.2123.12j8.20.0...0.0...1c.1.12.serp.yK8JLKTBibE also, there's some code in the video-media-bug branch you can record directly into mp4. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, May 16, 2013 at 6:28 AM, I put the Who? in Mishehu wrote: > For multimedia format conversion needs, I generally recommend ffmpeg. Please see http://www.ffmpeg.org/ > > It's a little more complex but you can script it very easily. Otherwise if you use KDE you can try out http://www.kdenlive.org/ for editing and transcoding. > > -Yossi > > On 05/15/2013 02:26 PM, D D wrote: > > Hi, > > > > I am using H264 codec for a call into FS, and recording it to an FSV file. > > > > How can I convert the FSV file to MP4? > > > > Any ideas on how to get started on this task would be greatly appreciated! > > > > Thanks, > > David > > > > > > > > > > > > > > _________________________________________________________________________ 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://wiki.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://wiki.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/20130516/324b9162/attachment-0001.html From sirimmfs at gmail.com Thu May 16 10:11:34 2013 From: sirimmfs at gmail.com (Siri MM) Date: Thu, 16 May 2013 16:11:34 +1000 Subject: [Freeswitch-users] IP change detected -> no internal profile In-Reply-To: References: <15534432.5814.1368605789796.JavaMail.javamailuser@localhost> <23490192.5864.1368605939908.JavaMail.javamailuser@localhost> Message-ID: Hi Michael, Thanks for the reply. Is 'auto-restart' suitable in a DHCP scenario? I may not know the IP that the device is going to acquire, or change, hence the question. Thanks! On Thu, May 16, 2013 at 3:08 PM, Michael Collins wrote: > > > > On Wed, May 15, 2013 at 5:33 PM, Siri MM wrote: > >> Hey Ray, >> >> Thanks for the response. I agree that this is a strange scenario - there >> seems to be some problem with the DHCP server in the environmnet, because >> of which, after lease time expiry, it sometimes doesn't provide back the IP >> to the device. I assume this triggers a interface down scenario in the >> device, causing the " loose ethX and FS to all available interfaces" >> scenario. I have a couple of questions though: >> 1. " loose ethX and you are binding FS to all available interfaces" - Is >> FS binding to available interfaces part configurable? Can I ask freeswitch >> not to bind to loop back interface if eth0 goes down? >> > I believe another poster mentioned setting the SIP profile param > 'auto-restart' to false. Try that first. > >> 2. Keeping in mind that this is a strange scenario, should FS still >> behave this way? I mean, once the device has got back the right IP, I >> wonder why the internal profile has not been loaded back? >> > This is indeed interesting. Does your ethX port actually get 127.0.0.1? Or > does Sofia just somehow end up on the lo interface? In any event, if you > disable the auto-restart feature it will prevent FS from bouncing over to > 127.0.0.1. > > -Michael > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/57a7bc78/attachment.html From ehermouet at bluetel.fr Thu May 16 13:38:48 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 16 May 2013 11:38:48 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> Message-ID: <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> hi i have this on my log 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 i'm sure i don't have configure my xml file but where and how ? yes provider use 2833 tks Le 2013-05-15 23:42, Michael Collins a ?crit?: > I recommend that you get a console debug log of the call and put it > up > on pastebin.freeswitch.org [11]. Hopefully well be able to see what > is > happening. > > In the meantime, is your provider set to use RFC2833 digits or do > they want to see in-band DTMFs? > -MC > > On Wed, May 15, 2013 at 2:01 PM, Erwan Hermouet [12]> wrote: > >> Hi all. We have voip provider connectd to FS and 1 extension. When >> my extension make call using voip provider dtmf not work. Hi phone >> to voip provider and dtmf is active to rfc2833. >> >> ? >> >> So i suppose i miss something on my config. Can you explain where?? >> tks advance >> >> ? >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [1] >> http://www.freeswitchsolutions.com [2] >> >> >> [3] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [4] >> http://wiki.freeswitch.org [5] >> http://www.cluecon.com [6] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [7] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [9] >> http://www.freeswitch.org [10] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [13] > http://www.ClueCon.com [14] > http://www.OSTAG.org [15] > > > > Links: > ------ > [1] mailto:consulting at freeswitch.org > [2] http://www.freeswitchsolutions.com > [3] > [4] http://www.freeswitch.org > [5] http://wiki.freeswitch.org > [6] http://www.cluecon.com > [7] mailto:FreeSWITCH-users at lists.freeswitch.org > [8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [9] http://lists.freeswitch.org/mailman/options/freeswitch-users > [10] http://www.freeswitch.org > [11] http://pastebin.freeswitch.org > [12] mailto:ehermouet at bluetel.fr > [13] http://www.FreeSWITCH.org > [14] http://www.ClueCon.com > [15] http://www.OSTAG.org From fernando at cloudigit.com Thu May 16 11:40:42 2013 From: fernando at cloudigit.com (Fernando Hernandez) Date: Thu, 16 May 2013 15:40:42 +0800 Subject: [Freeswitch-users] New codec integration and concurrent calls limit questions Message-ID: Hello everyone, we are working on a videoconferece platform with freeSWITCH integrated. Now we have a the possibility to try a new codec, proprietary, and we have never gone too deep with freeSWITCH. I have tried to take a look at the wiki but I didn?t find the answer to the next question, Is it possible to integrate that new codec into freeSWITCH? according to the codec provider it is a G.279 implementation, with several improvements. Any licence issue? difficulty to achieve the integration? On the other hand, we also have another issue. I have checked freeSWITCH wiki trying to find any possible limit for concurrent calls, and according to it, it doesn't seem to be a limit of SIP but purely the RTP. But with the open source video conference platform we are working now this is limited to 25, and according to the developers it is due to the audio. Do you have any idea about this issue? We have run some test and we go over 25 people in the meeting freeSWITCH uses 125% of CPU (ref. 8 cores machine and each core represent 100% capacity => 800%), is this a normal behavior? Thank you very much, Fernando Hernandez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/5555a482/attachment.html From idokan at gmail.com Thu May 16 18:44:25 2013 From: idokan at gmail.com (ik) Date: Thu, 16 May 2013 17:44:25 +0300 Subject: [Freeswitch-users] Freeswitch create bad SIP header Message-ID: Hello, I'm new to freeswitch, and I did google this, but can't find an answer that works for me. I have created the following configuration (both at Asterisk and FS sides): https://gist.github.com/ik5/1a0767f69b8065e1c8b8 * The IP addresses are hidden and replaced with "asterisk.example.com" and " freeswitch.example.com" As you can see there with the SIP header, freeswitch create the "From" and "To" field with the same IP/Domain address, and there is no "contact" field to make it contact properly. So what am I missing or doing wrong ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/19a60b43/attachment.html From shahzad.bhatti at g-r-v.com Thu May 16 19:51:06 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Thu, 16 May 2013 20:51:06 +0500 Subject: [Freeswitch-users] mod_cdr_csv is not add cdr data in csv file Message-ID: Hi everyone, i am using *FreeSWITCH Version 1.5.1b+git~20130515T182820Z~37d109b107 (git 37d109b 2013-05-15 18:28:20Z)* on CentOS and for the mod_cdr_csv my conf. is as: but the issue is that when i use *cdr_csv rotate* on fs_cli then sometime it not works and also the cdr data is not added in the csv file sometimes any suggestions is highly appreciated Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/381e03cd/attachment-0001.html From msc at freeswitch.org Thu May 16 20:18:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 May 2013 09:18:16 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> Message-ID: On Thu, May 16, 2013 at 2:38 AM, wrote: > hi > > i have this on my log > > 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf > receive payload to 101 > 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send > payload to 101 > > i'm sure i don't have configure my xml file but where and how ? > yes provider use 2833 > Can you pastebin the entire call from start to finish? Use pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax highlighting. Reply to this email thread with the URL from pastebin. -MC > > tks > > > Le 2013-05-15 23:42, Michael Collins a ?crit : > > I recommend that you get a console debug log of the call and put it > > up > > on pastebin.freeswitch.org [11]. Hopefully well be able to see what > > is > > happening. > > > > In the meantime, is your provider set to use RFC2833 digits or do > > they want to see in-band DTMFs? > > -MC > > > > On Wed, May 15, 2013 at 2:01 PM, Erwan Hermouet > [12]> wrote: > > > >> Hi all. We have voip provider connectd to FS and 1 extension. When > >> my extension make call using voip provider dtmf not work. Hi phone > >> to voip provider and dtmf is active to rfc2833. > >> > >> > >> > >> So i suppose i miss something on my config. Can you explain where ? > >> tks advance > >> > >> > >> > > > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org [1] > >> http://www.freeswitchsolutions.com [2] > >> > >> > >> [3] > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org [4] > >> http://wiki.freeswitch.org [5] > >> http://www.cluecon.com [6] > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org [7] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [9] > >> http://www.freeswitch.org [10] > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org [13] > > http://www.ClueCon.com [14] > > http://www.OSTAG.org [15] > > > > > > > > Links: > > ------ > > [1] mailto:consulting at freeswitch.org > > [2] http://www.freeswitchsolutions.com > > [3] > > [4] http://www.freeswitch.org > > [5] http://wiki.freeswitch.org > > [6] http://www.cluecon.com > > [7] mailto:FreeSWITCH-users at lists.freeswitch.org > > [8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [9] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [10] http://www.freeswitch.org > > [11] http://pastebin.freeswitch.org > > [12] mailto:ehermouet at bluetel.fr > > [13] http://www.FreeSWITCH.org > > [14] http://www.ClueCon.com > > [15] http://www.OSTAG.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/9ab0bf9f/attachment.html From mishehu at freeswitch.org Thu May 16 20:30:53 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 16 May 2013 11:30:53 -0500 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> <51940C18.10603@freeswitch.org> Message-ID: <519509BD.6040000@freeswitch.org> Heh, lysdexia strikes again! I read it as "FLV" not "FSV"... -Yossi On 05/16/2013 12:43 AM, Seven Du wrote: > fsv is not supported in standard ffmpeg, someone did some work on this > and possibly in the freeswitch_contrib repo, also you may find it by > following this: > > https://www.google.com.hk/search?safe=strict&rlz=1C5CHFA_enCN506CN506&q=lists.freeswitch.org+fsvdec.c&oq=lists.freeswitch.org+fsvdec.c&gs_l=serp.3...8140.13246.0.13663.21.21.0.0.0.3.153.2123.12j8.20.0...0.0...1c.1.12.serp.yK8JLKTBibE > > also, there's some code in the video-media-bug branch you can record > directly into mp4. > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Thursday, May 16, 2013 at 6:28 AM, I put the Who? in Mishehu wrote: > >> For multimedia format conversion needs, I generally recommend >> ffmpeg. Please see http://www.ffmpeg.org/ >> >> It's a little more complex but you can script it very easily. >> Otherwise if you use KDE you can try out http://www.kdenlive.org/ for >> editing and transcoding. >> >> -Yossi >> >> On 05/15/2013 02:26 PM, D D wrote: >>> Hi, >>> >>> I am using H264 codec for a call into FS, and recording it to an FSV >>> file. >>> >>> How can I convert the FSV file to MP4? >>> >>> Any ideas on how to get started on this task would be greatly >>> appreciated! >>> >>> Thanks, >>> David >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/2c6cc50a/attachment-0001.html From mishehu at freeswitch.org Thu May 16 20:44:01 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 16 May 2013 11:44:01 -0500 Subject: [Freeswitch-users] New codec integration and concurrent calls limit questions In-Reply-To: References: Message-ID: <51950CD1.7060408@freeswitch.org> See inline responses. On 05/16/2013 02:40 AM, Fernando Hernandez wrote: > Hello everyone, > > we are working on a videoconferece platform with freeSWITCH > integrated. Now we have a the possibility to try a new codec, > proprietary, and we have never gone too deep with freeSWITCH. I have > tried to take a look at the wiki but I didn?t find the answer to the > next question, > > Is it possible to integrate that new codec into freeSWITCH? according > to the codec provider it is a G.279 implementation, with several > improvements. Any licence issue? difficulty to achieve the integration? > Yes, it is possible. You have to look at the code for another codec's module for basic reference. Yes, there *can* be licensing issues, especially if you wish to distribute the code in any fashion. You need to read both the MPL 1.1 as well as the license terms for whatever libraries you are linking your new module with. But before I go any further, the issue with g729 isn't that the source code is necessarily restrictively licensed - it's that the codec itself is protected under patents and thus subject to per-channel licensing. I do find myself curious to know what this implementation of g729 that you are planning to work with is, and what are the improvements that it offers over the licensed g729 in the FreeSWITCH tree. I am curious in part because I know the guys put in a lot of effort to be able to provide legal licensing for g729 in FreeSWITCH, and if there's a viable way to improve on the code, there might be the option to provide a bounty for it or make the contribution yourself back to the FreeSWITCH code (without the use of the proprietary code, of course). > On the other hand, we also have another issue. I have checked > freeSWITCH wiki trying to find any possible limit for concurrent > calls, and according to it,it doesn't seem to be a limit of SIP but > purely the RTP. But with the open source video conference platform we > are working now this is limited to 25, and according to the developers > it is due to the audio. Do you have any idea about this issue? We have > run some test and we go over 25 people in the meeting freeSWITCH uses > 125% of CPU (ref. 8 cores machine and each core represent 100% > capacity => 800%), is this a normal behavior? Are you transcoding audio on these calls? That's where most of the CPU power would go to if the audio is indeed the issue. > > Thank you very much, > > Fernando Hernandez > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/d8aa584d/attachment.html From lesley.pervis at gmail.com Thu May 16 21:25:03 2013 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Thu, 16 May 2013 11:25:03 -0600 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: <517167CE.4010505@freeswitch.org> References: <517167CE.4010505@freeswitch.org> Message-ID: So I thought I'd try 1.2.9 mod_portaudio on RPi. Fresh install, running FS as root, with the only change to add mod_portaudio to modules.conf.xml, but mod_portaudio is unable to find devices. Alsa is working fine. I can aplay files and get output. Any ideas on how to get a somewhat stable branch of FS working with mod_portaudio? On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler < intralanman at freeswitch.org> wrote: > On 13-04-18 07:24 PM, Lesley Pervis wrote: > > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi > > seem made for each other. > > > > Forgot to mention, I was building on the stable branch. > we're testing the latest version of portaudio in a branch right now. > feel free to test and comment your findings on > http://jira.freeswitch.org/browse/FS-3387 > > -Ray > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/1c685968/attachment.html From ehermouet at bluetel.fr Thu May 16 21:29:44 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Thu, 16 May 2013 19:29:44 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> Message-ID: <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> I have the log but i never found how works pastebin ?? do you have tutorial ? tks De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : jeudi 16 mai 2013 18:18 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Thu, May 16, 2013 at 2:38 AM, wrote: hi i have this on my log 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf receive payload to 101 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf send payload to 101 i'm sure i don't have configure my xml file but where and how ? yes provider use 2833 Can you pastebin the entire call from start to finish? Use pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax highlighting. Reply to this email thread with the URL from pastebin. -MC tks Le 2013-05-15 23:42, Michael Collins a ?crit : > I recommend that you get a console debug log of the call and put it > up > on pastebin.freeswitch.org [11]. Hopefully well be able to see what > is > happening. > > In the meantime, is your provider set to use RFC2833 digits or do > they want to see in-band DTMFs? > -MC > > On Wed, May 15, 2013 at 2:01 PM, Erwan Hermouet [12]> wrote: > >> Hi all. We have voip provider connectd to FS and 1 extension. When >> my extension make call using voip provider dtmf not work. Hi phone >> to voip provider and dtmf is active to rfc2833. >> >> >> >> So i suppose i miss something on my config. Can you explain where ? >> tks advance >> >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [1] >> http://www.freeswitchsolutions.com [2] >> >> >> [3] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [4] >> http://wiki.freeswitch.org [5] >> http://www.cluecon.com [6] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [7] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [9] >> http://www.freeswitch.org [10] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [13] > http://www.ClueCon.com [14] > http://www.OSTAG.org [15] > > > > Links: > ------ > [1] mailto:consulting at freeswitch.org > [2] http://www.freeswitchsolutions.com > [3] > [4] http://www.freeswitch.org > [5] http://wiki.freeswitch.org > [6] http://www.cluecon.com > [7] mailto:FreeSWITCH-users at lists.freeswitch.org > [8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [9] http://lists.freeswitch.org/mailman/options/freeswitch-users > [10] http://www.freeswitch.org > [11] http://pastebin.freeswitch.org > [12] mailto:ehermouet at bluetel.fr > [13] http://www.FreeSWITCH.org > [14] http://www.ClueCon.com > [15] http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/a9f2507b/attachment-0001.html From lesley.pervis at gmail.com Thu May 16 21:42:14 2013 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Thu, 16 May 2013 11:42:14 -0600 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: <517167CE.4010505@freeswitch.org> References: <517167CE.4010505@freeswitch.org> Message-ID: Raymond, I think I found the right commit for your branch. I'll give it a whirl on the Pi. b514f62ed6c00949f407fda64abbf643681af4ee refs/heads/FS-3387_new_pa On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler < intralanman at freeswitch.org> wrote: > On 13-04-18 07:24 PM, Lesley Pervis wrote: > > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi > > seem made for each other. > > > > Forgot to mention, I was building on the stable branch. > we're testing the latest version of portaudio in a branch right now. > feel free to test and comment your findings on > http://jira.freeswitch.org/browse/FS-3387 > > -Ray > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/8add1e71/attachment.html From sertys at gmail.com Thu May 16 21:42:41 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 16 May 2013 19:42:41 +0200 Subject: [Freeswitch-users] Sdp manipulation help In-Reply-To: References: Message-ID: I am having trouble putting srtp to optional mode. UAs send sdp with rtp/avp and a:crypto lines which im permitting. But i want to decide whether the call uses srtp or zrtp. The UAs send also zrtp-hash. I export sip_secure_media to false but am still seeing the a:crypto lines being sent to the b-leg. I cut them off in my dialplan with set switch_r_sdp, but that breaks the whole negotiation, because the sdp isn't properly patched afterwards for nat( i also set a bunch of other vars afterwards). Is there a polite way of telling fs to remove the a:crypto lines and break the srtp and thus activating zrtp. Running 1.2.8git with inbound_zrtp_passthru to true. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/87293247/attachment.html From jmesquita at freeswitch.org Thu May 16 21:52:08 2013 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 16 May 2013 14:52:08 -0300 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: References: <517167CE.4010505@freeswitch.org> Message-ID: I've chasing this rabbit for some time but latest pa fails to compile on MacOSX Montain Lion.... That's a bitch Jo?o Mesquita FreeSWITCH? Solutions On Thu, May 16, 2013 at 2:42 PM, Lesley Pervis wrote: > Raymond, I think I found the right commit for your branch. I'll give it a > whirl on the Pi. > > b514f62ed6c00949f407fda64abbf643681af4ee refs/heads/FS-3387_new_pa > > > On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler < > intralanman at freeswitch.org> wrote: > >> On 13-04-18 07:24 PM, Lesley Pervis wrote: >> > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi >> > seem made for each other. >> > >> > Forgot to mention, I was building on the stable branch. >> we're testing the latest version of portaudio in a branch right now. >> feel free to test and comment your findings on >> http://jira.freeswitch.org/browse/FS-3387 >> >> -Ray >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/97885167/attachment.html From krice at freeswitch.org Thu May 16 22:16:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 16 May 2013 13:16:08 -0500 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: Message-ID: It works... You need the oss kernel mod loaded in the kernel, and you need asound dev packages loaded then rebuild... Your USB device should then show up in pa devlist I have some info on switchpi.org reguarding running FS on RaspPi On 5/16/13 12:25 PM, "Lesley Pervis" wrote: > So I thought I'd try 1.2.9 mod_portaudio on RPi. Fresh install, running FS as > root, with the only change to add mod_portaudio to modules.conf.xml, but > mod_portaudio is unable to find devices. Alsa is working fine. I can aplay > files and get output. Any ideas on how to get a somewhat stable branch of FS > working with mod_portaudio? > > > On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler > wrote: >> On 13-04-18 07:24 PM, Lesley Pervis wrote: >>> > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi >>> > seem made for each other. >>> > >>> > Forgot to mention, I was building on the stable branch. >> we're testing the latest version of portaudio in a branch right now. >> feel free to test and comment your findings on >> http://jira.freeswitch.org/browse/FS-3387 >> >> -Ray >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/c8f0d094/attachment.html From sdame at 207me.com Thu May 16 22:37:08 2013 From: sdame at 207me.com (Stephen Dame) Date: Thu, 16 May 2013 14:37:08 -0400 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: References: Message-ID: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> Fernando, if it?s BigBluebutton your referring to a couple of things. Since it is currently a flash client, you are limited to codecs you can use. BBB is using speex/16 which takes more cpu cycles to mix in mod conference. Getting 8-10 speex users mixing takes up a standard core, so you can project how it scales based on hardware thrown at it. Also red5 is used for streaming video, etc, and this needs cpu cycles. There are no limits to BBB in software, and you can run very much larger conference than 25 if you scale the hardware appropriately. You can even distribute the app across multiple servers. Getting a proprietary codec from client to FreeSWITCH would be a challenge. The <25 in BBB forums is intended to set expectations for user setting up standard single server implementations. Regards, Stephen Hostbbb.com From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- users-bounces at lists.freeswitch.org] On Behalf Of Fernando Hernandez Sent: Thursday, May 16, 2013 3:41 AM To: freeswitch-users at lists.freeswitch.org Cc: ???; ??? Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions Hello everyone, we are working on a videoconferece platform with freeSWITCH integrated. Now we have a the possibility to try a new codec, proprietary, and we have never gone too deep with freeSWITCH. I have tried to take a look at the wiki but I didn?t find the answer to the next question, Is it possible to integrate that new codec into freeSWITCH? according to the codec provider it is a G.279 implementation, with several improvements. Any licence issue? difficulty to achieve the integration? On the other hand, we also have another issue. I have checked freeSWITCH wiki trying to find any possible limit for concurrent calls, and according to it, it doesn't seem to be a limit of SIP but purely the RTP. But with the open source video conference platform we are working now this is limited to 25, and according to the developers it is due to the audio. Do you have any idea about this issue? We have run some test and we go over 25 people in the meeting freeSWITCH uses 125% of CPU (ref. 8 cores machine and each core represent 100% capacity => 800%), is this a normal behavior? Thank you very much, Fernando Hernandez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/9ba91a84/attachment-0001.html From lesley.pervis at gmail.com Thu May 16 23:15:15 2013 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Thu, 16 May 2013 13:15:15 -0600 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: References: Message-ID: Great info, thanks. Once I've got it working, I'll add a Raspberry Pi section to the wiki with the exact steps to get it working. In the meantime, I think this page more or less explains what you're talking about. http://wiki.debian.org/SoundFAQ In summary, "apt-get install libasound2-dev alsa-oss" before compiling and either load the OSS modules by hand with "modprobe snd_pcm_oss" and "modprobe snd_mixer_oss" or add the OSS modules to /etc/modules so they load at boot. You said on your voipusersconference appearance that you're not cross compiling. Still true? There's sure not much oomph to a Pi. On Thu, May 16, 2013 at 12:16 PM, Ken Rice wrote: > It works... You need the oss kernel mod loaded in the kernel, and you > need asound dev packages loaded then rebuild... Your USB device should then > show up in pa devlist > > I have some info on switchpi.org reguarding running FS on RaspPi > > > > On 5/16/13 12:25 PM, "Lesley Pervis" wrote: > > So I thought I'd try 1.2.9 mod_portaudio on RPi. Fresh install, running FS > as root, with the only change to add mod_portaudio to modules.conf.xml, but > mod_portaudio is unable to find devices. Alsa is working fine. I can aplay > files and get output. Any ideas on how to get a somewhat stable branch of > FS working with mod_portaudio? > > > On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler < > intralanman at freeswitch.org> wrote: > > On 13-04-18 07:24 PM, Lesley Pervis wrote: > > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi > > seem made for each other. > > > > Forgot to mention, I was building on the stable branch. > we're testing the latest version of portaudio in a branch right now. > feel free to test and comment your findings on > http://jira.freeswitch.org/browse/FS-3387 > > -Ray > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/28e9d0be/attachment.html From krice at freeswitch.org Thu May 16 23:25:03 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 16 May 2013 14:25:03 -0500 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: References: Message-ID: <0AD85490-D0D2-4037-848B-903378F3B21E@freeswitch.org> yep no cross compiling... it takes a while lol... looking to get maybe the same platform the the raspbian guys use for building packages Ken Sent from my iPad On May 16, 2013, at 14:15, Lesley Pervis wrote: > Great info, thanks. > > Once I've got it working, I'll add a Raspberry Pi section to the wiki with the exact steps to get it working. > > In the meantime, I think this page more or less explains what you're talking about. > > http://wiki.debian.org/SoundFAQ > > In summary, "apt-get install libasound2-dev alsa-oss" before compiling and either load the OSS modules by hand with "modprobe snd_pcm_oss" and "modprobe snd_mixer_oss" or add the OSS modules to /etc/modules so they load at boot. > > You said on your voipusersconference appearance that you're not cross compiling. Still true? There's sure not much oomph to a Pi. > > > On Thu, May 16, 2013 at 12:16 PM, Ken Rice wrote: >> It works... You need the oss kernel mod loaded in the kernel, and you need asound dev packages loaded then rebuild... Your USB device should then show up in pa devlist >> >> I have some info on switchpi.org reguarding running FS on RaspPi >> >> >> >> On 5/16/13 12:25 PM, "Lesley Pervis" wrote: >> >> So I thought I'd try 1.2.9 mod_portaudio on RPi. Fresh install, running FS as root, with the only change to add mod_portaudio to modules.conf.xml, but mod_portaudio is unable to find devices. Alsa is working fine. I can aplay files and get output. Any ideas on how to get a somewhat stable branch of FS working with mod_portaudio? >> >> >> On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler wrote: >> On 13-04-18 07:24 PM, Lesley Pervis wrote: >> > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi >> > seem made for each other. >> > >> > Forgot to mention, I was building on the stable branch. >> we're testing the latest version of portaudio in a branch right now. >> feel free to test and comment your findings on >> http://jira.freeswitch.org/browse/FS-3387 >> >> -Ray >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/64a3ff09/attachment-0001.html From d at d-man.org Fri May 17 01:18:32 2013 From: d at d-man.org (Darren Schreiber) Date: Thu, 16 May 2013 14:18:32 -0700 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: <5194270F.20407@hkcradio.com> Message-ID: Since Francis is now in charge of the project, I'll ask him to do it :-) Then you guys will get to know him as well. He's very sharp and will not really have anything but this project to work on for a while, and phone provisioning. So, short answer, yes, I'll be happy to :-) - Darren From: Nick Giannak > Organization: HKC Radio Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Wednesday, May 15, 2013 5:23 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX Darren, Can you do us a favor and post here when it's fixed, so we can play with it again? You once recommended Kazoo to me but it was so far beyond the scope of what I needed or knew what to do with. It's nice to see blue.box back in development! Thanks, Nick On 5/15/2013 3:30 PM, Darren Schreiber wrote: Hi Mehroz, There is an issue with the current git and nobody here has had the time to fix it, but the ISO works. That said, the project has become dated, I agree. It's definitely not dead though. We are actually rebuilding both blue.box items this week (the ISO and fixing git), which should fix the build. It's definitely not dead. As of last week, Francis has been dedicated full-time on blue.box. You will see activity on it via that route. You can also always ask questions in #2600hz. - Darren From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Tuesday, May 14, 2013 11:44 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/eb43642e/attachment.html From j at 2600hz.com Fri May 17 01:25:49 2013 From: j at 2600hz.com (Joshua Goldbard) Date: Thu, 16 May 2013 21:25:49 +0000 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: Looping Francis in so you guys can yell at him belligerently :D. Cheers, Joshua Joshua Goldbard VP of Marketing, 2600hz 116 Natoma Street, Floor 2 San Francisco, CA, 94104 415.886.7923 | j at 2600hz.com On May 16, 2013, at 2:18 PM, Darren Schreiber > wrote: Since Francis is now in charge of the project, I'll ask him to do it :-) Then you guys will get to know him as well. He's very sharp and will not really have anything but this project to work on for a while, and phone provisioning. So, short answer, yes, I'll be happy to :-) - Darren From: Nick Giannak > Organization: HKC Radio Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Wednesday, May 15, 2013 5:23 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX Darren, Can you do us a favor and post here when it's fixed, so we can play with it again? You once recommended Kazoo to me but it was so far beyond the scope of what I needed or knew what to do with. It's nice to see blue.box back in development! Thanks, Nick On 5/15/2013 3:30 PM, Darren Schreiber wrote: Hi Mehroz, There is an issue with the current git and nobody here has had the time to fix it, but the ISO works. That said, the project has become dated, I agree. It's definitely not dead though. We are actually rebuilding both blue.box items this week (the ISO and fixing git), which should fix the build. It's definitely not dead. As of last week, Francis has been dedicated full-time on blue.box. You will see activity on it via that route. You can also always ask questions in #2600hz. - Darren From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Tuesday, May 14, 2013 11:44 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-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://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/20130516/1180bba4/attachment.html From andretodd at verizon.net Fri May 17 01:57:03 2013 From: andretodd at verizon.net (Andre) Date: Thu, 16 May 2013 17:57:03 -0400 Subject: [Freeswitch-users] remove Message-ID: <071f01ce5280$4f4c8730$ede59590$@verizon.net> Hi, can you remove me from the distro? It's just too many emails. I'll add myself back later when I'm ready to use freeswitch. Thank you, Andre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/6584c29c/attachment-0001.html From jkomar at jbox.ca Fri May 17 02:02:20 2013 From: jkomar at jbox.ca (Komar, Jason) Date: Thu, 16 May 2013 16:02:20 -0600 Subject: [Freeswitch-users] remove In-Reply-To: <071f01ce5280$4f4c8730$ede59590$@verizon.net> References: <071f01ce5280$4f4c8730$ede59590$@verizon.net> Message-ID: There is an unsubscribe link at the bottom of each email that comes from the list. On Thu, May 16, 2013 at 3:57 PM, Andre wrote: > Hi, can you remove me from the distro? It?s just too many emails.**** > > ** ** > > I?ll add myself back later when I?m ready to use freeswitch.**** > > ** ** > > Thank you,**** > > Andre**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/eee69596/attachment.html From msc at freeswitch.org Fri May 17 02:12:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 May 2013 15:12:37 -0700 Subject: [Freeswitch-users] Freeswitch create bad SIP header In-Reply-To: References: Message-ID: What are you trying to accomplish? If you just want to send unauth'd calls to/from each box that doesn't require a gateway in FreeSWITCH. However, if you want the FreeSWITCH box to register to the Asterisk box then you'll need a gateway. If you just want to send a call from FreeSWITCH to Asterisk then you can use a simple dialstring in your bridge line: -MC On Thu, May 16, 2013 at 7:44 AM, ik wrote: > Hello, > > I'm new to freeswitch, and I did google this, but can't find an answer > that works for me. > > I have created the following configuration (both at Asterisk and FS sides): > https://gist.github.com/ik5/1a0767f69b8065e1c8b8 > > * The IP addresses are hidden and replaced with "asterisk.example.com" > and "freeswitch.example.com" > > As you can see there with the SIP header, freeswitch create the "From" and > "To" field with the same IP/Domain address, and there is no "contact" field > to make it contact properly. > > So what am I missing or doing wrong ? > > Thanks, > > Ido > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/c344c40d/attachment.html From msc at freeswitch.org Fri May 17 02:16:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 May 2013 15:16:38 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> Message-ID: On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet wrote: > I have the log but i never found how works pastebin ?? do you have > tutorial ? > There isn't a tutorial. You log on, paste your stuff into the text box, select FreeSWITCH Log as the syntax highlighting and then click Send. Copy the URL from the browse address bar. it will be something like: http://pastebin.freeswitch.org/20927 -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/645d28c7/attachment.html From sdevoy at bizfocused.com Fri May 17 02:31:48 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 16 May 2013 18:31:48 -0400 Subject: [Freeswitch-users] One Way Audio Message-ID: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/84d1dd9e/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/84d1dd9e/attachment-0001.gif From jaganthoutam at gmail.com Fri May 17 03:24:27 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Thu, 16 May 2013 19:24:27 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue Message-ID: HI, i have a freeswitch as a VOIP Gateway that will handle the Inbound and Outbound calls with media proxy, here issue comes My Freeswith is Sending FreeSwitch as a callid Number.. and callid name as a Number i wondered why it is sending like that i even comment the effective acller id name and caller id but still same issue, can any one help on this. Version : FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Thanks Jagan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/29f8982d/attachment.html From msc at freeswitch.org Fri May 17 03:35:13 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 16 May 2013 16:35:13 -0700 Subject: [Freeswitch-users] One Way Audio In-Reply-To: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> Message-ID: Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: > Hi all,**** > > ** ** > > First, I am on version: FreeSWITCH Version > 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) > I hope to move to the Stable 1.2.9 this weekend.**** > > ** ** > > I am having very reliable one way audio when calling Sprint Cell Phone > users, though not every time. I got this reproducible on my phone, but by > the time I learned tcpdump command, it was working again. However, the > user that reported it seems to get it almost everytime.**** > > ** ** > > Helpful tidbits:**** > > **? **I THINK it happens in either direction.**** > > **? **For this person at his home, it appears to be every time > (for now)**** > > **? **He reports calling (to or from) other Sprint Cell users > results in the same problem from our FS**** > > **? **It appears to only be true with Sprint Cell calls! (But my > users say that?s not Sprints fault!)**** > > ** ** > > Scenario:**** > > I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and > it rings, he answers, I can hear him crystal clear ? he can?t hear me at > all. I had a theory that it worked after 30 seconds (on my cell), but that > does not hold true on his cell.**** > > ** ** > > Here is the FS logfile: > http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** > ** > > And here is the tcpdump output: > http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip **** > > ** ** > > Based on the small size of the file, I suspect someone is going to say ?do > it again with this tcpdump command?. I welcome the education.**** > > ** ** > > Anyway, any insight will be appreciated.**** > > ** ** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/f25936a1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/f25936a1/attachment.gif From josuputa at yahoo.es Fri May 17 03:33:20 2013 From: josuputa at yahoo.es (JoZu) Date: Thu, 16 May 2013 16:33:20 -0700 (PDT) Subject: [Freeswitch-users] Skype 4.1 skypopen compatible? Message-ID: <1368747200787-7590784.post@n2.nabble.com> I read this in new Skype 4.1 client for linux: Implemented support for secondary logins Now you can run more than one instance of Skype on Linux simultaneously. Just add the ?? secondary? command line option to launch a second client. Its possible to implement this new option in skypopen with Sebastian trick? PD: Read skypopen wiki and the Skype 4.0 ! (unsupported) by Sebastian Fiorentini http://wiki.freeswitch.org/wiki/Skypopen#Skype_4.0_.21_.28unsupported.29 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-4-1-skypopen-compatible-tp7590784.html Sent from the freeswitch-users mailing list archive at Nabble.com. From info at shishir.com.np Fri May 17 04:53:34 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Thu, 16 May 2013 17:53:34 -0700 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Try on profile setting. On 16.05.2013 16:24, Jagadish Thoutam wrote: > HI, > ? > > ?i have a freeswitch as a VOIP Gateway that will handle the Inbound > and Outbound calls with media proxy, here issue comes My Freeswith is > Sending FreeSwitch as a callid Number.. ? ? ? ? ? ? ? ? and > callid name as a Number ? ?? i wondered why it is sending like that > i even comment the? effective acller id name and caller id but still > same issue, can any one help on this. > > Version ? : > > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git > e1c325d > 2013-04-23 19:49:07Z) > > Thanks > Jagan ?? From dujinfang at gmail.com Fri May 17 05:52:31 2013 From: dujinfang at gmail.com (Seven Du) Date: Fri, 17 May 2013 09:52:31 +0800 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: <519509BD.6040000@freeswitch.org> References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> <51940C18.10603@freeswitch.org> <519509BD.6040000@freeswitch.org> Message-ID: Ha, again FLV is not supported in FS, but I was working on a mod_ffmpeg which potentially work for anything, it is far from complete but I think it could record sth. :) On Friday, May 17, 2013 at 12:30 AM, I put the Who? in Mishehu wrote: > Heh, lysdexia strikes again! I read it as "FLV" not "FSV"... > > -Yossi > > On 05/16/2013 12:43 AM, Seven Du wrote: > > fsv is not supported in standard ffmpeg, someone did some work on this and possibly in the freeswitch_contrib repo, also you may find it by following this: > > > > https://www.google.com.hk/search?safe=strict&rlz=1C5CHFA_enCN506CN506&q=lists.freeswitch.org+fsvdec.c&oq=lists.freeswitch.org+fsvdec.c&gs_l=serp.3...8140.13246.0.13663.21.21.0.0.0.3.153.2123.12j8.20.0...0.0...1c.1.12.serp.yK8JLKTBibE > > > > also, there's some code in the video-media-bug branch you can record directly into mp4. > > > > -- > > Seven Du > > http://www.freeswitch.org.cn > > http://about.me/dujinfang > > http://www.dujinfang.com > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > On Thursday, May 16, 2013 at 6:28 AM, I put the Who? in Mishehu wrote: > > > > > For multimedia format conversion needs, I generally recommend ffmpeg. Please see http://www.ffmpeg.org/ > > > > > > It's a little more complex but you can script it very easily. Otherwise if you use KDE you can try out http://www.kdenlive.org/ for editing and transcoding. > > > > > > -Yossi > > > > > > On 05/15/2013 02:26 PM, D D wrote: > > > > Hi, > > > > > > > > I am using H264 codec for a call into FS, and recording it to an FSV file. > > > > > > > > How can I convert the FSV file to MP4? > > > > > > > > Any ideas on how to get started on this task would be greatly appreciated! > > > > > > > > Thanks, > > > > David > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ 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://wiki.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://wiki.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://wiki.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://wiki.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/20130517/3b25eb34/attachment-0001.html From william.king at quentustech.com Fri May 17 06:19:30 2013 From: william.king at quentustech.com (William King) Date: Thu, 16 May 2013 19:19:30 -0700 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> <51940C18.10603@freeswitch.org> <519509BD.6040000@freeswitch.org> Message-ID: <519593B2.1040604@quentustech.com> Have you tried to play the FLV with VLC player? if that works, the audio could work with mod_vlc. 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 05/16/2013 06:52 PM, Seven Du wrote: > Ha, again FLV is not supported in FS, but I was working on a mod_ffmpeg > which potentially work for anything, it is far from complete but I think > it could record sth. :) > > On Friday, May 17, 2013 at 12:30 AM, I put the Who? in Mishehu wrote: > >> Heh, lysdexia strikes again! I read it as "FLV" not "FSV"... >> >> -Yossi >> >> On 05/16/2013 12:43 AM, Seven Du wrote: >>> fsv is not supported in standard ffmpeg, someone did some work on >>> this and possibly in the freeswitch_contrib repo, also you may find >>> it by following this: >>> >>> https://www.google.com.hk/search?safe=strict&rlz=1C5CHFA_enCN506CN506&q=lists.freeswitch.org+fsvdec.c&oq=lists.freeswitch.org+fsvdec.c&gs_l=serp.3...8140.13246.0.13663.21.21.0.0.0.3.153.2123.12j8.20.0...0.0...1c.1.12.serp.yK8JLKTBibE >>> >>> >>> also, there's some code in the video-media-bug branch you can record >>> directly into mp4. >>> >>> -- >>> Seven Du >>> http://www.freeswitch.org.cn >>> http://about.me/dujinfang >>> http://www.dujinfang.com >>> >>> Sent with Sparrow >>> >>> On Thursday, May 16, 2013 at 6:28 AM, I put the Who? in Mishehu wrote: >>> >>>> For multimedia format conversion needs, I generally recommend >>>> ffmpeg. Please see http://www.ffmpeg.org/ >>>> >>>> It's a little more complex but you can script it very easily. >>>> Otherwise if you use KDE you can try out http://www.kdenlive.org/ >>>> for editing and transcoding. >>>> >>>> -Yossi >>>> >>>> On 05/15/2013 02:26 PM, D D wrote: >>>>> Hi, >>>>> >>>>> I am using H264 codec for a call into FS, and recording it to an >>>>> FSV file. >>>>> >>>>> How can I convert the FSV file to MP4? >>>>> >>>>> Any ideas on how to get started on this task would be greatly >>>>> appreciated! >>>>> >>>>> Thanks, >>>>> David >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From sdevoy at bizfocused.com Fri May 17 06:20:00 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 16 May 2013 22:20:00 -0400 Subject: [Freeswitch-users] One Way Audio In-Reply-To: References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> Message-ID: <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/47e171f9/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130516/47e171f9/attachment.gif From mishehu at freeswitch.org Fri May 17 06:27:21 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 16 May 2013 21:27:21 -0500 Subject: [Freeswitch-users] How can I convert an FSV file to MP4? In-Reply-To: References: <1368460459.40937.YahooMailNeo@web120705.mail.ne1.yahoo.com> <1368645978.66700.YahooMailNeo@web120703.mail.ne1.yahoo.com> <51940C18.10603@freeswitch.org> <519509BD.6040000@freeswitch.org> Message-ID: <51959589.3010009@freeswitch.org> Me just being pedantic - actually 1 module in FreeSWITCH does use FLV format at least for audio: mod_rtmp. However, Mathie Rene's code currently allows for speex packets although current versions of Flash support also g711u. :-) -Yossi On 05/16/2013 08:52 PM, Seven Du wrote: > Ha, again FLV is not supported in FS, but I was working on a > mod_ffmpeg which potentially work for anything, it is far from > complete but I think it could record sth. :) > > On Friday, May 17, 2013 at 12:30 AM, I put the Who? in Mishehu wrote: > >> Heh, lysdexia strikes again! I read it as "FLV" not "FSV"... >> >> -Yossi >> >> On 05/16/2013 12:43 AM, Seven Du wrote: >>> fsv is not supported in standard ffmpeg, someone did some work on >>> this and possibly in the freeswitch_contrib repo, also you may find >>> it by following this: >>> >>> https://www.google.com.hk/search?safe=strict&rlz=1C5CHFA_enCN506CN506&q=lists.freeswitch.org+fsvdec.c&oq=lists.freeswitch.org+fsvdec.c&gs_l=serp.3...8140.13246.0.13663.21.21.0.0.0.3.153.2123.12j8.20.0...0.0...1c.1.12.serp.yK8JLKTBibE >>> >>> also, there's some code in the video-media-bug branch you can record >>> directly into mp4. >>> >>> -- >>> Seven Du >>> http://www.freeswitch.org.cn >>> http://about.me/dujinfang >>> http://www.dujinfang.com >>> >>> Sent with Sparrow >>> >>> On Thursday, May 16, 2013 at 6:28 AM, I put the Who? in Mishehu wrote: >>> >>>> For multimedia format conversion needs, I generally recommend >>>> ffmpeg. Please see http://www.ffmpeg.org/ >>>> >>>> It's a little more complex but you can script it very easily. >>>> Otherwise if you use KDE you can try out http://www.kdenlive.org/ >>>> for editing and transcoding. >>>> >>>> -Yossi >>>> >>>> On 05/15/2013 02:26 PM, D D wrote: >>>>> Hi, >>>>> >>>>> I am using H264 codec for a call into FS, and recording it to an >>>>> FSV file. >>>>> >>>>> How can I convert the FSV file to MP4? >>>>> >>>>> Any ideas on how to get started on this task would be greatly >>>>> appreciated! >>>>> >>>>> Thanks, >>>>> David >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/96016e51/attachment-0001.html From fernando at cloudigit.com Fri May 17 06:54:55 2013 From: fernando at cloudigit.com (Fernando Hernandez) Date: Fri, 17 May 2013 10:54:55 +0800 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> References: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> Message-ID: Thanks a lot to both for your advice, it has been really useful. Stephen, yes, I am referring to BBB. And I have a question, as far as BBB is not limited by software and we need to run conference larger than 25 we need to scale the hardware. But, I would like to know about the possibility to distribute BBB across multiple servers, how can we do it? do you know if there is any documentation about it? And about Misheshu comments, I think the licence issue will be a problem. The implementation of G.729 I am talking about is proprietary, not open at all, so we need to find out if it is possible to use it with FreeSWITCH (I am not well versed on the licensing issues, so I will read the licenses and try to understand them). You can check this comparison table to have an idea about this codec performance, [image: Im?genes integradas 1] And finally, about audio transcoding. We use BBB and speex for the encoding, but as Stephen said before, maybe the audio is not the only problem we have. We need to run more tests... Thank you very much and best regards, Fernando Hernandez. 2013/5/17 Stephen Dame > Fernando, if it?s BigBluebutton your referring to a couple of things.*** > * > > ** ** > > Since it is currently a flash client, you are limited to codecs you can > use. BBB is using speex/16 which takes more cpu cycles to mix in mod > conference. Getting 8-10 speex users mixing takes up a standard core, > so you can project how it scales based on hardware thrown at it.**** > > ** ** > > Also red5 is used for streaming video, etc, and this needs cpu cycles.**** > > ** ** > > There are no limits to BBB in software, and you can run very much larger > conference than 25 if you scale the hardware appropriately. You can even > distribute the app across multiple servers.**** > > ** ** > > Getting a proprietary codec from client to FreeSWITCH would be a challenge. > **** > > ** ** > > The <25 in BBB forums is intended to set expectations for user setting up > standard single server implementations.**** > > ** ** > > Regards,**** > > Stephen**** > > Hostbbb.com**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Fernando > Hernandez > *Sent:* Thursday, May 16, 2013 3:41 AM > *To:* freeswitch-users at lists.freeswitch.org > *Cc:* ???; ??? > *Subject:* [Freeswitch-users] New codec integration and concurrent calls > limitquestions**** > > ** ** > > Hello everyone,**** > > ** ** > > we are working on a videoconferece platform with freeSWITCH integrated. > Now we have a the possibility to try a new codec, proprietary, and we have > never gone too deep with freeSWITCH. I have tried to take a look at the > wiki but I didn?t find the answer to the next question,**** > > ** ** > > Is it possible to integrate that new codec into freeSWITCH? according to > the codec provider it is a G.279 implementation, with several improvements. > Any licence issue? difficulty to achieve the integration? **** > > ** ** > > On the other hand, we also have another issue. I have checked freeSWITCH > wiki trying to find any possible limit for concurrent calls, and according > to it, it doesn't seem to be a limit of SIP but purely the RTP. But with > the open source video conference platform we are working now this is > limited to 25, and according to the developers it is due to the audio. Do > you have any idea about this issue? We have run some test and we go over 25 > people in the meeting freeSWITCH uses 125% of CPU (ref. 8 cores machine and > each core represent 100% capacity => 800%), is this a normal behavior?**** > > ** ** > > Thank you very much,**** > > ** ** > > Fernando Hernandez**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/2190ba26/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 69805 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/2190ba26/attachment-0001.png From mishehu at freeswitch.org Fri May 17 07:16:55 2013 From: mishehu at freeswitch.org (I put the Who? in Mishehu) Date: Thu, 16 May 2013 22:16:55 -0500 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: References: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> Message-ID: <5195A127.3010309@freeswitch.org> Unfortunately the table you provided doesn't explain adequately what it is that I'm looking at a comparison of. I'm a bit confused because g729 does not support all of the sample rates here, and according to what I've read g729.1 supports wideband up to 32k. If you could please clarify what these statistics are for, I'd appreciate it. I am not able to wrap my head around what the proprietary implementation of g729 that you are referring to (you have not revealed what it is) does that the existing code does not do or could not be made to do. -Yossi From jaganthoutam at gmail.com Fri May 17 07:32:11 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Thu, 16 May 2013 23:32:11 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: No, it didn't help me Still i am getting FreeSWITH as a caller ID (not Caller ID Name) caller id Name i am getting Number. Thanks Jagan On 16 May 2013 20:53, wrote: > Try on profile setting. > > On 16.05.2013 16:24, Jagadish Thoutam wrote: > > HI, > > > > > > i have a freeswitch as a VOIP Gateway that will handle the Inbound > > and Outbound calls with media proxy, here issue comes My Freeswith is > > Sending FreeSwitch as a callid Number.. and > > callid name as a Number i wondered why it is sending like that > > i even comment the effective acller id name and caller id but still > > same issue, can any one help on this. > > > > Version : > > > > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git > > e1c325d > > 2013-04-23 19:49:07Z) > > > > Thanks > > Jagan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130516/18385d51/attachment.html From alex at opensystems.net.au Fri May 17 09:24:17 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Fri, 17 May 2013 13:24:17 +0800 Subject: [Freeswitch-users] Freeswitch & Opsview Monitoring Message-ID: I thought I'd post this for people reference. I've configured opsview (Nagios) to monitor my current sessions & graph it so I can match performance & usage. On the Freeswitch server install the opsview-agent & set service to auto start Create check_freeswitch_sessions.sh in /usr/local/nagios/libexec & chkmod +x the script -------------------------------------------------------- #!/bin/bash set fsStatus=" " /usr/local/freeswitch/bin/fs_cli -x "show calls count" > /tmp/fsStatus if [ $? -ne 0 ]; then echo "Critical: Freeswitch not responding!" rm /tmp/fsStatus exit -1 fi fsCalls=`grep "total." IT Consultant for Open Systems Support www.opensystems.net.au -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/1eeefdae/attachment.html From fernando at cloudigit.com Fri May 17 09:32:47 2013 From: fernando at cloudigit.com (Fernando Hernandez) Date: Fri, 17 May 2013 13:32:47 +0800 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: <5195A127.3010309@freeswitch.org> References: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> <5195A127.3010309@freeswitch.org> Message-ID: I will try to send more specific information. But, at this moment, i don't have it. I am almost sure you doesn't know about this implementation. It is from an Asian University. Sorry if I can not give more details. Fernando. 2013/5/17 I put the Who? in Mishehu > Unfortunately the table you provided doesn't explain adequately what it > is that I'm looking at a comparison of. I'm a bit confused because g729 > does not support all of the sample rates here, and according to what > I've read g729.1 supports wideband up to 32k. If you could please > clarify what these statistics are for, I'd appreciate it. I am not able > to wrap my head around what the proprietary implementation of g729 that > you are referring to (you have not revealed what it is) does that the > existing code does not do or could not be made to do. > > -Yossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/19c92af5/attachment.html From ehermouet at bluetel.fr Fri May 17 09:46:00 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Fri, 17 May 2013 07:46:00 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> Message-ID: <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> On login i try my email...but don t work...i loose here Michael Collins a ?crit?: >On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >wrote: > >> I have the log but i never found how works pastebin ?? do you have >> tutorial ? >> >There isn't a tutorial. You log on, paste your stuff into the text box, >select FreeSWITCH Log as the syntax highlighting and then click Send. >Copy >the URL from the browse address bar. it will be something like: >http://pastebin.freeswitch.org/20927 > >-MC > >-- >Michael S Collins >Twitter: @mercutioviz >http://www.FreeSWITCH.org >http://www.ClueCon.com >http://www.OSTAG.org > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/7b8ee52a/attachment-0001.html From lesley.pervis at gmail.com Fri May 17 10:06:40 2013 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Fri, 17 May 2013 00:06:40 -0600 Subject: [Freeswitch-users] mod_portaudio on Raspberry Pi currently broken? In-Reply-To: <0AD85490-D0D2-4037-848B-903378F3B21E@freeswitch.org> References: <0AD85490-D0D2-4037-848B-903378F3B21E@freeswitch.org> Message-ID: Thanks for the pointers, Ken, I got it working on the stable branch. I'll add the info to the mod_portaudio wiki page when my wiki account application gets approved. Found a great Ars Technica article on the Raspbian package building you mentioned. http://arstechnica.com/information-technology/2013/03/how-two-volunteers-built-the-raspberry-pis-operating-system/ On Thu, May 16, 2013 at 1:25 PM, Ken Rice wrote: > yep no cross compiling... it takes a while lol... looking to get maybe the > same platform the the raspbian guys use for building packages > > Ken > Sent from my iPad > > On May 16, 2013, at 14:15, Lesley Pervis wrote: > > Great info, thanks. > > Once I've got it working, I'll add a Raspberry Pi section to the wiki with > the exact steps to get it working. > > In the meantime, I think this page more or less explains what you're > talking about. > > http://wiki.debian.org/SoundFAQ > > In summary, "apt-get install libasound2-dev alsa-oss" before compiling and > either load the OSS modules by hand with "modprobe snd_pcm_oss" and > "modprobe snd_mixer_oss" or add the OSS modules to /etc/modules so they > load at boot. > > You said on your voipusersconference appearance that you're not cross > compiling. Still true? There's sure not much oomph to a Pi. > > > On Thu, May 16, 2013 at 12:16 PM, Ken Rice wrote: > >> It works... You need the oss kernel mod loaded in the kernel, and you >> need asound dev packages loaded then rebuild... Your USB device should then >> show up in pa devlist >> >> I have some info on switchpi.org reguarding running FS on RaspPi >> >> >> >> On 5/16/13 12:25 PM, "Lesley Pervis" wrote: >> >> So I thought I'd try 1.2.9 mod_portaudio on RPi. Fresh install, running >> FS as root, with the only change to add mod_portaudio to modules.conf.xml, >> but mod_portaudio is unable to find devices. Alsa is working fine. I can >> aplay files and get output. Any ideas on how to get a somewhat stable >> branch of FS working with mod_portaudio? >> >> >> On Fri, Apr 19, 2013 at 9:50 AM, Raymond Chandler < >> intralanman at freeswitch.org> wrote: >> >> On 13-04-18 07:24 PM, Lesley Pervis wrote: >> > Wow, that's great, thanks. Hadn't heard of switchpi yet. FS and pi >> > seem made for each other. >> > >> > Forgot to mention, I was building on the stable branch. >> we're testing the latest version of portaudio in a branch right now. >> feel free to test and comment your findings on >> http://jira.freeswitch.org/browse/FS-3387 >> >> -Ray >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/84a70324/attachment.html From john at telecube.com.au Fri May 17 10:16:55 2013 From: john at telecube.com.au (Telecube - John) Date: Fri, 17 May 2013 16:16:55 +1000 Subject: [Freeswitch-users] Freeswitch & Opsview Monitoring In-Reply-To: References: Message-ID: <5195CB57.6000801@telecube.com.au> Nice, if it's ok with you I'm going to 'borrow' this and make it work with Cactus - John On 17/05/13 3:24 PM, Alex Ynema wrote: > I thought I'd post this for people reference. I've configured opsview > (Nagios) to monitor my current sessions & graph it so I can match > performance & usage. > > On the Freeswitch server install the opsview-agent & set service to > auto start > > Create check_freeswitch_sessions.sh in /usr/local/nagios/libexec & > chkmod +x the script > -------------------------------------------------------- > #!/bin/bash > > set fsStatus=" " > /usr/local/freeswitch/bin/fs_cli -x "show calls count" > /tmp/fsStatus > if [ $? -ne 0 ]; then > echo "Critical: Freeswitch not responding!" > rm /tmp/fsStatus > exit -1 > fi > > fsCalls=`grep "total." > #Strip unwanted chars from the channels responses to get pure ints. > fsCalls=${fsCalls/ total./} > > # Build a message up > message="Active Sessions: $fsCalls" > echo $message > > rm /tmp/fsStatus > #exit $fsCalls > exit 0 > -------------------------------------------------------- > > The add the following to /usr/local/nagios/nrpe.cfg > > -------------------------------------------------------- > command[check_freeswitch_sessions]=/usr/local/nagios/libexec/check_freeswitch_Sessions.sh > -------------------------------------------------------- > > On Opsview server use check_nrpe service check to remotely run the > check_freeswitch_sessions service check. > > Then on your opsview server > > edit /usr/local/nagios/etc/map and add the following > -------------------------------------------------------- > # Service type: Freeswitch Sessions > # ouput:Active Sessions: 0 > /output:Active.*?(\d+)/ > and push @s, [ "session", > [ "sessions", GAUGE, $1 ] ]; > -------------------------------------------------------- > > *Alex Ynema***| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/0e1f9d25/attachment-0001.html From smrdoshi at gmail.com Fri May 17 08:38:21 2013 From: smrdoshi at gmail.com (smrdoshi) Date: Thu, 16 May 2013 21:38:21 -0700 (PDT) Subject: [Freeswitch-users] opensips & Freeswitch calling issue Message-ID: <1368765501780-7590795.post@n2.nabble.com> Hi, I have follow below steps to implement load balancing of FS servers using opensips. http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs Here opensips is forwarding registration request to FS servers. But I need users to register only in opensips and when they do calls, FS should work as media server. So in simple, opensips should work as user registration server and Freeswitch should work as media server. I did some modifications in opensips.cfg to do registration in opensips only. Here is the modifications, Removed below code from script if (!add_path_received()) { xlog("L_ERR", "$ci|log|unable to add path"); sl_send_reply("503", "Internal path befuddlement"); #remove the association between the call-id and the media server (if one) #but leave the contact user and server to support transfers cache_remove("local", "$ci"); xlog("L_INFO", "$ci|end|cleaned up call id from cache"); exit; } Added code in place of above code if (!zz_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if ( proto==TCP || 0 ) setflag(7); if (!save("location")) sl_reply_error(); exit; Using above code registration is working fine. But now the problem is with DID number calls. I have two extensions registered in opensips 12345 and 123456. I have DID number 0123456789 in FS. >From Freeswitch I have pointed DID to 123456 extension using sofia/profile/123456 at 192.168.1.10. Here, 192.168.1.10 is opensips ip 192.168.1.31 is fs ip But when i am calling that DID number, I am getting error in opensips like, http://pastebin.com/AhuDP6hC Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In Progress" I am trying trial & error and searching for solution but not getting anything. Please guys help me to fix the issue. Thanks in Advance. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795.html Sent from the freeswitch-users mailing list archive at Nabble.com. From alex at opensystems.net.au Fri May 17 11:15:57 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Fri, 17 May 2013 15:15:57 +0800 Subject: [Freeswitch-users] Freeswitch & Opsview Monitoring In-Reply-To: <5195CB57.6000801@telecube.com.au> References: <5195CB57.6000801@telecube.com.au> Message-ID: I already borrowed the script from someone here anyway so feel free to use it. If you come up with any good changes or additions reply with them here so others can use that too. Alex Ynema | IT Consultant alex at opensystems.net.au Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 17/05/2013 2:27 PM, "Telecube - John" wrote: > Nice, if it's ok with you I'm going to 'borrow' this and make it work > with Cactus > > - John > > On 17/05/13 3:24 PM, Alex Ynema wrote: > > I thought I'd post this for people reference. I've configured opsview > (Nagios) to monitor my current sessions & graph it so I can match > performance & usage. > > On the Freeswitch server install the opsview-agent & set service to auto > start > > Create check_freeswitch_sessions.sh in /usr/local/nagios/libexec & > chkmod +x the script > -------------------------------------------------------- > #!/bin/bash > > set fsStatus=" " > /usr/local/freeswitch/bin/fs_cli -x "show calls count" > /tmp/fsStatus > if [ $? -ne 0 ]; then > echo "Critical: Freeswitch not responding!" > rm /tmp/fsStatus > exit -1 > fi > > fsCalls=`grep "total." > #Strip unwanted chars from the channels responses to get pure ints. > fsCalls=${fsCalls/ total./} > > # Build a message up > message="Active Sessions: $fsCalls" > echo $message > > rm /tmp/fsStatus > #exit $fsCalls > exit 0 > -------------------------------------------------------- > > The add the following to /usr/local/nagios/nrpe.cfg > > -------------------------------------------------------- > > command[check_freeswitch_sessions]=/usr/local/nagios/libexec/check_freeswitch_Sessions.sh > -------------------------------------------------------- > > On Opsview server use check_nrpe service check to remotely run the > check_freeswitch_sessions service check. > > Then on your opsview server > > edit /usr/local/nagios/etc/map and add the following > -------------------------------------------------------- > # Service type: Freeswitch Sessions > # ouput:Active Sessions: 0 > /output:Active.*?(\d+)/ > and push @s, [ "session", > [ "sessions", GAUGE, $1 ] ]; > -------------------------------------------------------- > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130517/2dafb36c/attachment.html From krice at freeswitch.org Fri May 17 11:25:17 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 17 May 2013 02:25:17 -0500 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> Message-ID: <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> it tells you the password in the popup... this is an anti spam thing Ken Sent from my iPad On May 17, 2013, at 0:46, Hermouet Erwan wrote: > On login i try my email...but don t work...i loose here > > > Michael Collins a ?crit : >> >> >> >> >> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet wrote: >>> I have the log but i never found how works pastebin ?? do you have tutorial ? >>> >> There isn't a tutorial. You log on, paste your stuff into the text box, select FreeSWITCH Log as the syntax highlighting and then click Send. Copy the URL from the browse address bar. it will be something like: >> http://pastebin.freeswitch.org/20927 >> >> -MC > > Hermouet Erwan > Responsable technique > Bluetel > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/42e21274/attachment.html From ehermouet at bluetel.fr Fri May 17 11:44:13 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 17 May 2013 09:44:13 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> Message-ID: <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> i'm so stupid :) tks http://pastebin.freeswitch.org/20933 called num is 022206... and when i try to use dtmf touch 5 it's not works. tks Le 2013-05-17 09:25, Ken Rice a ?crit?: > it tells you the password in the popup... this is an anti spam thing > > KenSent from my iPad > > On May 17, 2013, at 0:46, Hermouet Erwan > wrote: > >> On login i try my email...but don t work...i loose here >> >> Michael Collins a ?crit : >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >>> wrote: >>> >>>> I have the log but i never found how works pastebin ?? do you >>>> have tutorial ? >>> >>> There isn't a tutorial. You log on, paste your stuff into the text >>> box, select FreeSWITCH Log as the syntax highlighting and then >>> click Send. Copy the URL from the browse address bar. it will be >>> something like: >>> http://pastebin.freeswitch.org/20927 [2] >>> >>> -MC >> >> Hermouet Erwan >> Responsable technique >> Bluetel > >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [4] >> http://www.freeswitchsolutions.com [5] >> >> >> [6] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [7] >> http://wiki.freeswitch.org [8] >> http://www.cluecon.com [9] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [10] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [11] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [12] >> http://www.freeswitch.org [13] > > > Links: > ------ > [1] mailto:ehermouet at bluetel.fr > [2] http://pastebin.freeswitch.org/20927 > [3] mailto:msc at freeswitch.org > [4] mailto:consulting at freeswitch.org > [5] http://www.freeswitchsolutions.com > [6] > [7] http://www.freeswitch.org > [8] http://wiki.freeswitch.org > [9] http://www.cluecon.com > [10] mailto:FreeSWITCH-users at lists.freeswitch.org > [11] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users > [13] http://www.freeswitch.org > [14] mailto:ehermouet at bluetel.fr From steveu at coppice.org Fri May 17 12:05:52 2013 From: steveu at coppice.org (Steve Underwood) Date: Fri, 17 May 2013 16:05:52 +0800 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: References: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> Message-ID: <5195E4E0.9090307@coppice.org> On 05/17/2013 10:54 AM, Fernando Hernandez wrote: > Thanks a lot to both for your advice, it has been really useful. > > Stephen, yes, I am referring to BBB. And I have a question, as far as > BBB is not limited by software and we need to run conference larger > than 25 we need to scale the hardware. But, I would like to know about > the possibility to distribute BBB across multiple servers, how can we > do it? do you know if there is any documentation about it? > > And about Misheshu comments, I think the licence issue will be a > problem. The implementation of G.729 I am talking about is > proprietary, not open at all, so we need to find out if it is possible > to use it with FreeSWITCH (I am not well versed on the licensing > issues, so I will read the licenses and try to understand them). > You can check this comparison table to have an idea about this codec > performance, > > Im?genes integradas 1 > > And finally, about audio transcoding. We use BBB and speex for the > encoding, but as Stephen said before, maybe the audio is not the only > problem we have. We need to run more tests... > > Thank you very much and best regards, > > Fernando Hernandez. What is this table supposed to indicate? u-law, A-law and most forms of ADPCM (the table doesn't indicate which one) are only designed for use at 8k samples/second, and give poor results when used at higher rates. G.729 can *only* be used at 8k samples/second. Its just useless at higher sample rates. If "capacity" is supposed to be the number of channels of the codec which can be supported on an i7-2720QM why is the number for G.729 so low? Even a poor implementation should be faster than 50 channels on a quad core processor like that. The G.729 for Freeswitch is certainly a lot faster than 148 channels on such a processor. Steve From khorsmann at gmail.com Fri May 17 12:26:42 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Fri, 17 May 2013 10:26:42 +0200 Subject: [Freeswitch-users] opensips & Freeswitch calling issue In-Reply-To: <1368765501780-7590795.post@n2.nabble.com> References: <1368765501780-7590795.post@n2.nabble.com> Message-ID: Hi, you should take a look with wireshark on the sip flow. It seems to be an endless loop. Your selfmade opensips xlog error messages didnt shows up whats going on. 2013/5/17 smrdoshi > Hi, > > I have follow below steps to implement load balancing of FS servers using > opensips. > > http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs > > Here opensips is forwarding registration request to FS servers. But I need > users to register only in opensips and when they do calls, FS should work > as > media server. So in simple, opensips should work as user registration > server > and Freeswitch should work as media server. > > I did some modifications in opensips.cfg to do registration in opensips > only. Here is the modifications, > > Removed below code from script > if (!add_path_received()) > { > xlog("L_ERR", "$ci|log|unable to add path"); > sl_send_reply("503", "Internal path befuddlement"); > > #remove the association between the call-id and the media server > (if one) > #but leave the contact user and server to support transfers > cache_remove("local", "$ci"); > xlog("L_INFO", "$ci|end|cleaned up call id from cache"); > exit; > } > Added code in place of above code > if (!zz_authorize("", "subscriber")) > { > www_challenge("", "0"); > exit; > } > > if (!db_check_to()) > { > sl_send_reply("403","Forbidden auth ID"); > exit; > } > > if ( proto==TCP || 0 ) setflag(7); > > if (!save("location")) > sl_reply_error(); > > exit; > > Using above code registration is working fine. > > But now the problem is with DID number calls. I have two extensions > registered in opensips 12345 and 123456. I have DID number 0123456789 in > FS. > >From Freeswitch I have pointed DID to 123456 extension using > sofia/profile/123456 at 192.168.1.10. > > Here, > 192.168.1.10 is opensips ip > 192.168.1.31 is fs ip > > But when i am calling that DID number, I am getting error in opensips like, > http://pastebin.com/AhuDP6hC > Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In > Progress" > > I am trying trial & error and searching for solution but not getting > anything. > > Please guys help me to fix the issue. > Thanks in Advance. > > Thanks, > Sam > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795.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://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 > -- Mit freundlichen Gr??en *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/f038d9bb/attachment.html From dvl36.ripe.nick at gmail.com Fri May 17 12:56:55 2013 From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko) Date: Fri, 17 May 2013 11:56:55 +0300 Subject: [Freeswitch-users] New codec integration and concurrent calls limitquestions In-Reply-To: <5195E4E0.9090307@coppice.org> References: <011a01ce5264$5fd4c570$1f7e5050$@207me.com> <5195E4E0.9090307@coppice.org> Message-ID: I think they mean modified CS-ACELP codec based on g.729. 50 simultaneously channels is low number on quad core I7, even using ITU reference implementation. 2013/5/17 Steve Underwood > On 05/17/2013 10:54 AM, Fernando Hernandez wrote: > > Thanks a lot to both for your advice, it has been really useful. > > > > Stephen, yes, I am referring to BBB. And I have a question, as far as > > BBB is not limited by software and we need to run conference larger > > than 25 we need to scale the hardware. But, I would like to know about > > the possibility to distribute BBB across multiple servers, how can we > > do it? do you know if there is any documentation about it? > > > > And about Misheshu comments, I think the licence issue will be a > > problem. The implementation of G.729 I am talking about is > > proprietary, not open at all, so we need to find out if it is possible > > to use it with FreeSWITCH (I am not well versed on the licensing > > issues, so I will read the licenses and try to understand them). > > You can check this comparison table to have an idea about this codec > > performance, > > > > Im?genes integradas 1 > > > > And finally, about audio transcoding. We use BBB and speex for the > > encoding, but as Stephen said before, maybe the audio is not the only > > problem we have. We need to run more tests... > > > > Thank you very much and best regards, > > > > Fernando Hernandez. > What is this table supposed to indicate? u-law, A-law and most forms of > ADPCM (the table doesn't indicate which one) are only designed for use > at 8k samples/second, and give poor results when used at higher rates. > G.729 can *only* be used at 8k samples/second. Its just useless at > higher sample rates. If "capacity" is supposed to be the number of > channels of the codec which can be supported on an i7-2720QM why is the > number for G.729 so low? Even a poor implementation should be faster > than 50 channels on a quad core processor like that. The G.729 for > Freeswitch is certainly a lot faster than 148 channels on such a processor. > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/0203a470/attachment.html From andrew at cassidywebservices.co.uk Fri May 17 13:04:09 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Fri, 17 May 2013 10:04:09 +0100 Subject: [Freeswitch-users] originate_timeout Message-ID: Hi guys, is it originate_timeout or origination_timeout? Both have almost identical wiki pages: http://wiki.freeswitch.org/wiki/Variable_originate_timeout http://wiki.freeswitch.org/wiki/Variable_origination_timeout -- *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/20130517/d80dfdb9/attachment.html From gmaruzz at gmail.com Fri May 17 15:52:41 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 17 May 2013 13:52:41 +0200 Subject: [Freeswitch-users] Skype 4.1 skypopen compatible? In-Reply-To: <1368747200787-7590784.post@n2.nabble.com> References: <1368747200787-7590784.post@n2.nabble.com> Message-ID: Hello Jozu at the moment is not possible, because the code works sending signaling to different X-servers. But, just for my information, what would be the advantage in using this "multiple client in same X server" feature? -giovanni On Fri, May 17, 2013 at 1:33 AM, JoZu wrote: > I read this in new Skype 4.1 client for linux: > > Implemented support for secondary logins > > Now you can run more than one instance of Skype on Linux simultaneously. > Just add the ?? secondary? command line option to launch a second client. > > Its possible to implement this new option in skypopen with Sebastian trick? > PD: Read skypopen wiki and the Skype 4.0 ! (unsupported) > by Sebastian Fiorentini > > http://wiki.freeswitch.org/wiki/Skypopen#Skype_4.0_.21_.28unsupported.29 > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Skype-4-1-skypopen-compatible-tp7590784.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://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 > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/ad769ead/attachment-0001.html From smrdoshi at gmail.com Fri May 17 13:07:32 2013 From: smrdoshi at gmail.com (smrdoshi) Date: Fri, 17 May 2013 02:07:32 -0700 (PDT) Subject: [Freeswitch-users] opensips & Freeswitch calling issue In-Reply-To: References: <1368765501780-7590795.post@n2.nabble.com> Message-ID: <1368781652733-7590807.post@n2.nabble.com> Hi Karsten, Thanks for the reply. I took tcpdump of FS sip log. Its showing 408 Request timeout error now. I am not sure where's the problem? I am attaching tcpdump in case if you get idea about the issue. tcpdump.pcap Please let me know. Thanks again for your help. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795p7590807.html Sent from the freeswitch-users mailing list archive at Nabble.com. From asilva at wirelessmundi.com Fri May 17 16:23:32 2013 From: asilva at wirelessmundi.com (Antonio Silva) Date: Fri, 17 May 2013 14:23:32 +0200 Subject: [Freeswitch-users] Re-Invite t38 is incomplete when t38_passtrough=true Message-ID: <1368793412.2854.116.camel@marces.madrid.commsmundi.com> Hi, i have the following scenario when sending a fax: ATA => FS => Provider => destination The ATA tries to send a fax to destination and in the middle of the call there is the re-invite in t38 from the provider. In dialplan i have: The invite from the provider is: INVITE sip:gw+TRUNKIP at 10.10.1.254:5060;transport=udp;gw=TRUNKIP SIP/2.0 Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bKno6j3r305os1nm4ph2a1sb0000010.1 Call-ID: badda7ee-3980-1231-8db4-b4b52f5e67f8 From: ;tag=dg2aai02-CC-52 To: "5313" ;tag=t6DB42KFXarmN CSeq: 2 INVITE Contact: Max-Forwards: 69 Content-Length: 437 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 38317113 38317116 IN IP4 10.10.1.2 s=Sip Call c=IN IP4 10.10.1.2 t=0 0 m=image 29090 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy m=audio 34264 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=silenceSupp:off - - - - a=ecan:fb on - a=X-fax a=fmtp:101 0-15 And the invite sent to the ATA is: INVITE sip:5313 at 192.168.1.97:5060;ref=5313 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.254;rport;branch=z9hG4bK87p0gpcv4p31g Max-Forwards: 70 From: ;tag=mKN5ZFeQ19XDr To: "5313" ;tag=5de97158a7c051b0o0 Call-ID: 6aa68878-1a412510 at 192.168.1.97 CSeq: 44056889 INVITE Contact: User-Agent: suvoz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 291 v=0 o=FreeSWITCH 1368772703 1368772705 IN IP4 10.10.1.254 s=FreeSWITCH c=IN IP4 10.10.1.254 t=0 0 m=image 13196 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:500 a=T38FaxMaxDatagram:500 a=T38FaxUdpEC:t38UDPRedundancy As you can see the audio part is not send to the ata. I understand that is another "call" between the ATA and FS, but should FS send an invite to the ATA containing a audio part as well? I'm not sure that is a bug or Is the normal behavior? should i open a jira for it? Searching for previous issues i found this, but there was no reply ... http://freeswitch-users.2379917.n2.nabble.com/Doubts-about-T38-passthru-td7582305.html Thanks for the help, Ant?nio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/459621b6/attachment.html From josuputa at yahoo.es Fri May 17 16:37:06 2013 From: josuputa at yahoo.es (JoZu) Date: Fri, 17 May 2013 05:37:06 -0700 (PDT) Subject: [Freeswitch-users] Skype 4.1 skypopen compatible? In-Reply-To: References: <1368747200787-7590784.post@n2.nabble.com> Message-ID: <1368794226624-7590811.post@n2.nabble.com> Maybe multiple Skypes with the same Skype username?. I thought that most modern versions of skype that was not possible. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-4-1-skypopen-compatible-tp7590784p7590811.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jason.holden at start.ca Fri May 17 17:00:58 2013 From: jason.holden at start.ca (Jason Holden) Date: Fri, 17 May 2013 09:00:58 -0400 Subject: [Freeswitch-users] server side features with Polycom soundpoint phones Message-ID: Does anyone have any documentation on this specificly for dnd and cf? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/10654f82/attachment.html From vipkilla at gmail.com Fri May 17 17:09:53 2013 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 17 May 2013 09:09:53 -0400 Subject: [Freeswitch-users] server side features with Polycom soundpoint phones In-Reply-To: References: Message-ID: Does anyone have any documentation on this specificly for dnd and cf?**** Currently FreeSWITCH does not support server controlled DND and CF. It's part of CSTA which is not implemented. I've made a beta module, mod_csta which can handle server controlled DND and CF but it requires some changes to mod_sofia which have not been implemented. Specifically the changed are in the SWITCH_EVENT_NOTIFY because currently that event is too limited to handle proper SIP NOTIFYs I'd be happy to share my beta module but it won't work unless you modify mod_sofia. I've written a patch to mod_sofia to fix SWITCH_EVENT_NOTIFY and submitted it to jira but have not gotten a response yet. Here is the jira: http://jira.freeswitch.org/browse/FS-5378 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/a9528aec/attachment.html From gmaruzz at gmail.com Fri May 17 17:20:32 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 17 May 2013 15:20:32 +0200 Subject: [Freeswitch-users] Skype 4.1 skypopen compatible? In-Reply-To: <1368794226624-7590811.post@n2.nabble.com> References: <1368747200787-7590784.post@n2.nabble.com> <1368794226624-7590811.post@n2.nabble.com> Message-ID: On Fri, May 17, 2013 at 2:37 PM, JoZu wrote: > Maybe multiple Skypes with the same Skype username?. > I thought that most modern versions of skype that was not possible. > This is possible since the beginning with the current method, no problem at all. Check the wiki page ( http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk ). Also the automatic installer gives you that possibility, and indeed is very popular. On the opposite, probably it would be not possible to have multiple instance with the same skypename in the same Xserver. -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/f635a7ee/attachment.html From idokan at gmail.com Fri May 17 16:43:37 2013 From: idokan at gmail.com (ik) Date: Fri, 17 May 2013 15:43:37 +0300 Subject: [Freeswitch-users] Freeswitch create bad SIP header In-Reply-To: References: Message-ID: On Fri, May 17, 2013 at 1:12 AM, Michael Collins wrote: > What are you trying to accomplish? If you just want to send unauth'd calls > to/from each box that doesn't require a gateway in FreeSWITCH. However, if > you want the FreeSWITCH box to register to the Asterisk box then you'll > need a gateway. > > If you just want to send a call from FreeSWITCH to Asterisk then you can > use a simple dialstring in your bridge line: > > > Will it work also vice versa, that is Asterisk can call FS like that ? Thanks > > -MC > > > On Thu, May 16, 2013 at 7:44 AM, ik wrote: > >> Hello, >> >> I'm new to freeswitch, and I did google this, but can't find an answer >> that works for me. >> >> I have created the following configuration (both at Asterisk and FS >> sides): >> https://gist.github.com/ik5/1a0767f69b8065e1c8b8 >> >> * The IP addresses are hidden and replaced with "asterisk.example.com" >> and "freeswitch.example.com" >> >> As you can see there with the SIP header, freeswitch create the "From" >> and "To" field with the same IP/Domain address, and there is no "contact" >> field to make it contact properly. >> >> So what am I missing or doing wrong ? >> >> Thanks, >> >> Ido >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/028492c3/attachment-0001.html From mbodbg at gmx.net Fri May 17 18:21:09 2013 From: mbodbg at gmx.net (mbo) Date: Fri, 17 May 2013 16:21:09 +0200 Subject: [Freeswitch-users] originate_timeout In-Reply-To: References: Message-ID: In the current version it's originate_ timeout. if you click on page http://wiki.freeswitch.org/wiki/Variable_origination_timeout on "history" you can see that there is already the advice (/* origination_timeout (soon to be originate_timeout ;)).Seems it was renamed in 2010. Markus Am 17.05.2013 um 11:04 schrieb Andrew Cassidy : > Hi guys, is it originate_timeout or origination_timeout? > > Both have almost identical wiki pages: > > http://wiki.freeswitch.org/wiki/Variable_originate_timeout > http://wiki.freeswitch.org/wiki/Variable_origination_timeout > > -- > 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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/2e293645/attachment.html From josuputa at yahoo.es Fri May 17 19:40:32 2013 From: josuputa at yahoo.es (JoZu) Date: Fri, 17 May 2013 08:40:32 -0700 (PDT) Subject: [Freeswitch-users] Skype 4.1 skypopen compatible? In-Reply-To: References: <1368747200787-7590784.post@n2.nabble.com> <1368794226624-7590811.post@n2.nabble.com> Message-ID: <1368805232276-7590817.post@n2.nabble.com> Giovanni Maruzzelli, thanks for all and thanks for your great job. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Skype-4-1-skypopen-compatible-tp7590784p7590817.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jaganthoutam at gmail.com Fri May 17 19:45:18 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Fri, 17 May 2013 11:45:18 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Any help on this issue. sent from samsung S3 On 17 May 2013 11:32, "Jagadish Thoutam" wrote: > No, it didn't help me Still i am getting FreeSWITH as a caller ID (not > Caller ID Name) caller id Name i am getting Number. > > > > Thanks > Jagan > > > On 16 May 2013 20:53, wrote: > >> Try on profile setting. >> >> On 16.05.2013 16:24, Jagadish Thoutam wrote: >> > HI, >> > >> > >> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >> > and Outbound calls with media proxy, here issue comes My Freeswith is >> > Sending FreeSwitch as a callid Number.. and >> > callid name as a Number i wondered why it is sending like that >> > i even comment the effective acller id name and caller id but still >> > same issue, can any one help on this. >> > >> > Version : >> > >> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >> > e1c325d >> > 2013-04-23 19:49:07Z) >> > >> > Thanks >> > Jagan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130517/1e92277e/attachment.html From jackal at cybershroud.net Fri May 17 20:45:57 2013 From: jackal at cybershroud.net (Carlos Flor) Date: Fri, 17 May 2013 12:45:57 -0400 Subject: [Freeswitch-users] No audio after resuming from hold Message-ID: I have 2 soft-clients registered to a FS box. One calls the other and the call is connected. Both parties can hear each other. Party A puts Party B on hold. The soft-client for each party indicates that the call is on hold. Party A takes the call off of hold. Both clients update their displays to indicate that the call is no longer on hold, however neither party can hear audio. This only happens when I have proxy_media set to true (this is a requirement because I am using zrtp and can't use trusted mitm). I have a very simple dialplan setup to test this scenario. I did several packet captures and watched the SDP packets and subsequent RTP packets. During the call, Phone A sends his audio port to FS, FS sends it's B-Leg audio port to Phone B, Phone B responds to FS with it's audio port, and FS finally responds to Phone A with it's A-Leg audio port. RTP packets flow between these ports and everything works. When the call is placed on hold and then taken off of hold, the same process happens, with the clients usually choosing their same ports as before, and FS choosing new ports for each leg. Each client then sends RTP packets to the newly chosen FS audio ports for their corresponding leg of the call. However, FS sends ICMP port unreachable packets back to each phone and no audio passes. I am happy to provide any information (logs, packet captures, etc...) if that will help, just tell me specifically what you would like. I'm running FreeSWITCH Version 1.5.1b+git~20130424T213558Z~eeeb4fa445 (git eeeb4fa 2013-04-24 21:35:58Z). Thanks for any help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/85bfab11/attachment.html From shaheryarkh at gmail.com Fri May 17 21:01:11 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Fri, 17 May 2013 19:01:11 +0200 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Well, if you want to have variable caller id number and name then use "effective_caller_id_name" and "effective_caller_id_number" channel variables in your dial plan. Beispiel, OR if you want to use a global outbound caller id name and number then you can modify following in vars.xml, Thank you. On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam wrote: > Any help on this issue. > > sent from samsung S3 > On 17 May 2013 11:32, "Jagadish Thoutam" wrote: > >> No, it didn't help me Still i am getting FreeSWITH as a caller ID (not >> Caller ID Name) caller id Name i am getting Number. >> >> >> >> Thanks >> Jagan >> >> >> On 16 May 2013 20:53, wrote: >> >>> Try on profile setting. >>> >>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>> > HI, >>> > >>> > >>> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >>> > and Outbound calls with media proxy, here issue comes My Freeswith is >>> > Sending FreeSwitch as a callid Number.. and >>> > callid name as a Number i wondered why it is sending like that >>> > i even comment the effective acller id name and caller id but still >>> > same issue, can any one help on this. >>> > >>> > Version : >>> > >>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>> > e1c325d >>> > 2013-04-23 19:49:07Z) >>> > >>> > Thanks >>> > Jagan >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/c89fc77e/attachment-0001.html From msc at freeswitch.org Fri May 17 21:38:34 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 17 May 2013 10:38:34 -0700 Subject: [Freeswitch-users] Freeswitch create bad SIP header In-Reply-To: References: Message-ID: On Fri, May 17, 2013 at 5:43 AM, ik wrote: > > On Fri, May 17, 2013 at 1:12 AM, Michael Collins wrote: > >> What are you trying to accomplish? If you just want to send unauth'd >> calls to/from each box that doesn't require a gateway in FreeSWITCH. >> However, if you want the FreeSWITCH box to register to the Asterisk box >> then you'll need a gateway. >> >> If you just want to send a call from FreeSWITCH to Asterisk then you can >> use a simple dialstring in your bridge line: >> >> >> > > Will it work also vice versa, that is Asterisk can call FS like that ? > For incoming calls to FreeSWITCH you will need to handle that in the dialplan. If you are not using authentication then you'll just need to route the inbound calls in your public context. Look in conf/dialplan/public.xml for an example of routing inbound calls to 1000-1019. You'll see that unauthenticated calls come in to the public context and if the destination is 1000-1019 it transfers them over to the default context where they (hopefully) get connected to the actual user. -MC > > Thanks > > >> >> -MC >> >> >> On Thu, May 16, 2013 at 7:44 AM, ik wrote: >> >>> Hello, >>> >>> I'm new to freeswitch, and I did google this, but can't find an answer >>> that works for me. >>> >>> I have created the following configuration (both at Asterisk and FS >>> sides): >>> https://gist.github.com/ik5/1a0767f69b8065e1c8b8 >>> >>> * The IP addresses are hidden and replaced with "asterisk.example.com" >>> and "freeswitch.example.com" >>> >>> As you can see there with the SIP header, freeswitch create the "From" >>> and "To" field with the same IP/Domain address, and there is no "contact" >>> field to make it contact properly. >>> >>> So what am I missing or doing wrong ? >>> >>> Thanks, >>> >>> Ido >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/6d0b6308/attachment.html From mikemol at gmail.com Fri May 17 22:01:49 2013 From: mikemol at gmail.com (Michael Mol) Date: Fri, 17 May 2013 14:01:49 -0400 Subject: [Freeswitch-users] Ekiga and feature codes Message-ID: <5196708D.7030402@gmail.com> One thing that is not at all clear to me is how to get a SIP client such as Ekiga to be able to use feature codes. For example, I have #300 for parking calls, and #301 for retrieving them...but I cannot for the life of me figure out how to get Ekiga access to those features. I've tried: 1. Putting the call on hold, and then using the dialpad. 2. Transferring the call to the feature code address (looks to the remote user like the call was cleared) 3. Dialing #301 from Ekiga results in me hitting an auto-attendant wishing me to dial my party's extension... I'm really not sure how to proceed... -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 555 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/438df9e9/attachment.bin From idokan at gmail.com Fri May 17 23:40:01 2013 From: idokan at gmail.com (ik) Date: Fri, 17 May 2013 22:40:01 +0300 Subject: [Freeswitch-users] Freeswitch create bad SIP header In-Reply-To: References: Message-ID: On Fri, May 17, 2013 at 8:38 PM, Michael Collins wrote: > > > > On Fri, May 17, 2013 at 5:43 AM, ik wrote: > >> >> On Fri, May 17, 2013 at 1:12 AM, Michael Collins wrote: >> >>> What are you trying to accomplish? If you just want to send unauth'd >>> calls to/from each box that doesn't require a gateway in FreeSWITCH. >>> However, if you want the FreeSWITCH box to register to the Asterisk box >>> then you'll need a gateway. >>> >>> If you just want to send a call from FreeSWITCH to Asterisk then you can >>> use a simple dialstring in your bridge line: >>> >>> >>> >> >> Will it work also vice versa, that is Asterisk can call FS like that ? >> > For incoming calls to FreeSWITCH you will need to handle that in the > dialplan. If you are not using authentication then you'll just need to > route the inbound calls in your public context. Look in > conf/dialplan/public.xml for an example of routing inbound calls to > 1000-1019. You'll see that unauthenticated calls come in to the public > context and if the destination is 1000-1019 it transfers them over to the > default context where they (hopefully) get connected to the actual user. > Thank you very much, finally I understand the state of mind. It looks like Asterisk works much harder then it should be, if you compare it :) > > -MC > > >> >> Thanks >> >> >>> >>> -MC >>> >>> >>> On Thu, May 16, 2013 at 7:44 AM, ik wrote: >>> >>>> Hello, >>>> >>>> I'm new to freeswitch, and I did google this, but can't find an answer >>>> that works for me. >>>> >>>> I have created the following configuration (both at Asterisk and FS >>>> sides): >>>> https://gist.github.com/ik5/1a0767f69b8065e1c8b8 >>>> >>>> * The IP addresses are hidden and replaced with "asterisk.example.com" >>>> and "freeswitch.example.com" >>>> >>>> As you can see there with the SIP header, freeswitch create the "From" >>>> and "To" field with the same IP/Domain address, and there is no "contact" >>>> field to make it contact properly. >>>> >>>> So what am I missing or doing wrong ? >>>> >>>> Thanks, >>>> >>>> Ido >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/9494ee3c/attachment-0001.html From bdfoster at davri.com Sat May 18 00:13:04 2013 From: bdfoster at davri.com (Brian Foster) Date: Fri, 17 May 2013 16:13:04 -0400 Subject: [Freeswitch-users] Freeswitch create bad SIP header In-Reply-To: References: Message-ID: Welcome to FreeSWITCH :) On May 17, 2013 3:47 PM, "ik" wrote: > > > > On Fri, May 17, 2013 at 8:38 PM, Michael Collins wrote: > >> >> >> >> On Fri, May 17, 2013 at 5:43 AM, ik wrote: >> >>> >>> On Fri, May 17, 2013 at 1:12 AM, Michael Collins wrote: >>> >>>> What are you trying to accomplish? If you just want to send unauth'd >>>> calls to/from each box that doesn't require a gateway in FreeSWITCH. >>>> However, if you want the FreeSWITCH box to register to the Asterisk box >>>> then you'll need a gateway. >>>> >>>> If you just want to send a call from FreeSWITCH to Asterisk then you >>>> can use a simple dialstring in your bridge line: >>>> >>>> >>>> >>> >>> Will it work also vice versa, that is Asterisk can call FS like that ? >>> >> For incoming calls to FreeSWITCH you will need to handle that in the >> dialplan. If you are not using authentication then you'll just need to >> route the inbound calls in your public context. Look in >> conf/dialplan/public.xml for an example of routing inbound calls to >> 1000-1019. You'll see that unauthenticated calls come in to the public >> context and if the destination is 1000-1019 it transfers them over to the >> default context where they (hopefully) get connected to the actual user. >> > > > Thank you very much, finally I understand the state of mind. It looks like > Asterisk works much harder then it should be, if you compare it :) > > >> >> -MC >> >> >>> >>> Thanks >>> >>> >>>> >>>> -MC >>>> >>>> >>>> On Thu, May 16, 2013 at 7:44 AM, ik wrote: >>>> >>>>> Hello, >>>>> >>>>> I'm new to freeswitch, and I did google this, but can't find an answer >>>>> that works for me. >>>>> >>>>> I have created the following configuration (both at Asterisk and FS >>>>> sides): >>>>> https://gist.github.com/ik5/1a0767f69b8065e1c8b8 >>>>> >>>>> * The IP addresses are hidden and replaced with "asterisk.example.com" >>>>> and "freeswitch.example.com" >>>>> >>>>> As you can see there with the SIP header, freeswitch create the "From" >>>>> and "To" field with the same IP/Domain address, and there is no "contact" >>>>> field to make it contact properly. >>>>> >>>>> So what am I missing or doing wrong ? >>>>> >>>>> Thanks, >>>>> >>>>> Ido >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/c759232d/attachment.html From jaganthoutam at gmail.com Sat May 18 00:26:40 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Fri, 17 May 2013 16:26:40 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Hi Muhammad, i disable the global outbound caller id, My setup is like this DID -----------> FreeSWITCH -------> AsteriskServer ASTERISK is my gateway in My Asterisk is getting FreeSWITCH as CallerID not evean as CallerNAme. Thanks Jagan On 17 May 2013 13:01, Muhammad Shahzad wrote: > Well, if you want to have variable caller id number and name then use > "effective_caller_id_name" and "effective_caller_id_number" channel > variables in your dial plan. Beispiel, > > > > > OR if you want to use a global outbound caller id name and number then you > can modify following in vars.xml, > > > > > Thank you. > > > > > On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam wrote: > >> Any help on this issue. >> >> sent from samsung S3 >> On 17 May 2013 11:32, "Jagadish Thoutam" wrote: >> >>> No, it didn't help me Still i am getting FreeSWITH as a caller ID (not >>> Caller ID Name) caller id Name i am getting Number. >>> >>> >>> >>> Thanks >>> Jagan >>> >>> >>> On 16 May 2013 20:53, wrote: >>> >>>> Try on profile setting. >>>> >>>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>>> > HI, >>>> > >>>> > >>>> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >>>> > and Outbound calls with media proxy, here issue comes My Freeswith is >>>> > Sending FreeSwitch as a callid Number.. and >>>> > callid name as a Number i wondered why it is sending like that >>>> > i even comment the effective acller id name and caller id but still >>>> > same issue, can any one help on this. >>>> > >>>> > Version : >>>> > >>>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>>> > e1c325d >>>> > 2013-04-23 19:49:07Z) >>>> > >>>> > Thanks >>>> > Jagan >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/952bda04/attachment-0001.html From michel.brabants at gmail.com Sat May 18 00:36:17 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 17 May 2013 22:36:17 +0200 Subject: [Freeswitch-users] Ekiga and feature codes In-Reply-To: <5196708D.7030402@gmail.com> References: <5196708D.7030402@gmail.com> Message-ID: I suppose it should send dtmf. Freeswitch as a b2b should take care of the rest. Op 17 mei 2013 20:05 schreef "Michael Mol" het volgende: > One thing that is not at all clear to me is how to get a SIP client such > as Ekiga to be able to use feature codes. > > For example, I have #300 for parking calls, and #301 for retrieving > them...but I cannot for the life of me figure out how to get Ekiga > access to those features. > > I've tried: > > 1. Putting the call on hold, and then using the dialpad. > 2. Transferring the call to the feature code address (looks to the > remote user like the call was cleared) > 3. Dialing #301 from Ekiga results in me hitting an auto-attendant > wishing me to dial my party's extension... > > I'm really not sure how to proceed... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/3dc25040/attachment.html From sdevoy at bizfocused.com Sat May 18 00:53:55 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 17 May 2013 16:53:55 -0400 Subject: [Freeswitch-users] Detecting Errors when dialing through multiple gateways Message-ID: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> Hi All, I have been having pretty good luck with redundancy by dialing through multiple gateways like: If gateway cheap fails, the call goes through on the expensive gateway. That is fine, but I would like an email notification of the failure on the cheap gateway. Is there any way to do that? The support group at the gateway vender is very good at correcting the problem when notified. I would love to send them an email of failures automatically. Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/d78e5d0c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/d78e5d0c/attachment.gif From bdfoster at davri.com Sat May 18 01:12:10 2013 From: bdfoster at davri.com (Brian Foster) Date: Fri, 17 May 2013 17:12:10 -0400 Subject: [Freeswitch-users] Detecting Errors when dialing through multiple gateways In-Reply-To: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> References: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> Message-ID: Lua scripts are really good at doing this, and if you are crafty enough you can even give them a failure description and supporting sip trace. That IMO would be the best option. I'm mobile right now so I can't write you an example but maybe others can chime in. - BDF On May 17, 2013 4:59 PM, "Sean Devoy" wrote: > Hi All,**** > > ** ** > > I have been having pretty good luck with redundancy by dialing through > multiple gateways like:**** > > **** > > ** ** > > If gateway cheap fails, the call goes through on the expensive gateway. > That is fine, but I would like an email notification of the failure on the > cheap gateway. Is there any way to do that? The support group at the > gateway vender is very good at correcting the problem when notified. I > would love to send them an email of failures automatically.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130517/5e2efc7c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/5e2efc7c/attachment.gif From shaheryarkh at gmail.com Sat May 18 02:24:11 2013 From: shaheryarkh at gmail.com (Muhammad Shahzad) Date: Sat, 18 May 2013 00:24:11 +0200 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: How are you disabling the global caller id? This won't work, (its a common mistake done by many users when dealing with X-PRE-PROCESS) You have to do something like this, Notice the space around ' - ' in X-PRE-PROCESS. Also you should set some appropriate caller id for outbound call in your dial plan, as already i told you in my previous email. Thank you. On Fri, May 17, 2013 at 10:26 PM, Jagadish Thoutam wrote: > Hi Muhammad, > > i disable the global outbound caller id, > > My setup is like this > > DID -----------> FreeSWITCH -------> AsteriskServer > > > > > expression="^(?:\+1)(88XXXXXXXX)$"> > > > > > > ASTERISK is my gateway in My Asterisk is getting FreeSWITCH as CallerID > not evean as CallerNAme. > > > > > > Thanks > Jagan > > > > > > On 17 May 2013 13:01, Muhammad Shahzad wrote: > >> Well, if you want to have variable caller id number and name then use >> "effective_caller_id_name" and "effective_caller_id_number" channel >> variables in your dial plan. Beispiel, >> >> >> >> >> OR if you want to use a global outbound caller id name and number then >> you can modify following in vars.xml, >> >> >> >> >> Thank you. >> >> >> >> >> On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam > > wrote: >> >>> Any help on this issue. >>> >>> sent from samsung S3 >>> On 17 May 2013 11:32, "Jagadish Thoutam" wrote: >>> >>>> No, it didn't help me Still i am getting FreeSWITH as a caller ID >>>> (not Caller ID Name) caller id Name i am getting Number. >>>> >>>> >>>> >>>> Thanks >>>> Jagan >>>> >>>> >>>> On 16 May 2013 20:53, wrote: >>>> >>>>> Try on profile setting. >>>>> >>>>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>>>> > HI, >>>>> > >>>>> > >>>>> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >>>>> > and Outbound calls with media proxy, here issue comes My Freeswith is >>>>> > Sending FreeSwitch as a callid Number.. and >>>>> > callid name as a Number i wondered why it is sending like that >>>>> > i even comment the effective acller id name and caller id but still >>>>> > same issue, can any one help on this. >>>>> > >>>>> > Version : >>>>> > >>>>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>>>> > e1c325d >>>>> > 2013-04-23 19:49:07Z) >>>>> > >>>>> > Thanks >>>>> > Jagan >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Mit freundlichen Gr??en >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Mit freundlichen Gr??en Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/f22c963c/attachment-0001.html From msc at freeswitch.org Sat May 18 02:52:29 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 17 May 2013 15:52:29 -0700 Subject: [Freeswitch-users] Ekiga and feature codes In-Reply-To: References: <5196708D.7030402@gmail.com> Message-ID: On Fri, May 17, 2013 at 1:36 PM, Michel Brabants wrote: > I suppose it should send dtmf. Freeswitch as a b2b should take care of the > rest. > Well, it depends. Is the sequence "#300" meant to reach FreeSWITCH, or is that supposed to be for Ekiga itself? Same with #301. Not sure about that. Does it require any configuration? (I'm not an Ekiga guy so I can't tell you.) OTOH, if you have the vanilla configs for FreeSWITCH you can just transfer the call to "6000" and it will do a valet park for you. -MC > Op 17 mei 2013 20:05 schreef "Michael Mol" het > volgende: > >> One thing that is not at all clear to me is how to get a SIP client such >> as Ekiga to be able to use feature codes. >> >> For example, I have #300 for parking calls, and #301 for retrieving >> them...but I cannot for the life of me figure out how to get Ekiga >> access to those features. >> >> I've tried: >> >> 1. Putting the call on hold, and then using the dialpad. >> 2. Transferring the call to the feature code address (looks to the >> remote user like the call was cleared) >> 3. Dialing #301 from Ekiga results in me hitting an auto-attendant >> wishing me to dial my party's extension... >> >> I'm really not sure how to proceed... >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/007c37aa/attachment.html From msc at freeswitch.org Sat May 18 03:16:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 17 May 2013 16:16:40 -0700 Subject: [Freeswitch-users] Detecting Errors when dialing through multiple gateways In-Reply-To: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> References: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> Message-ID: On Fri, May 17, 2013 at 1:53 PM, Sean Devoy wrote: > Hi All,**** > > ** ** > > I have been having pretty good luck with redundancy by dialing through > multiple gateways like:**** > > > I'm curious why your're doing simultaneous dial instead of sequential dial here. In any case, you can have multiple bridge apps instead of doing the comma-sep list: Try it out. Also, don't forget the standard disclaimer . -MC > **** > > ** ** > > If gateway cheap fails, the call goes through on the expensive gateway. > That is fine, but I would like an email notification of the failure on the > cheap gateway. Is there any way to do that? The support group at the > gateway vender is very good at correcting the problem when notified. I > would love to send them an email of failures automatically.**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/6c4c54dc/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/6c4c54dc/attachment.gif From msc at freeswitch.org Sat May 18 03:20:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 17 May 2013 16:20:54 -0700 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: On Fri, May 17, 2013 at 3:24 PM, Muhammad Shahzad wrote: > How are you disabling the global caller id? This won't work, (its a common > mistake done by many users when dealing with X-PRE-PROCESS) > > > > You have to do something like this, > > > FYI, we recommend not using XML comments when trying to disable a preprocess directive. Just do something like this: That will do the trick and will make the code easier to read. -MC > > Notice the space around ' - ' in X-PRE-PROCESS. > > Also you should set some appropriate caller id for outbound call in your > dial plan, as already i told you in my previous email. > > Thank you. > > > > > On Fri, May 17, 2013 at 10:26 PM, Jagadish Thoutam > wrote: > >> Hi Muhammad, >> >> i disable the global outbound caller id, >> >> My setup is like this >> >> DID -----------> FreeSWITCH -------> AsteriskServer >> >> >> >> >> > expression="^(?:\+1)(88XXXXXXXX)$"> >> >> >> >> >> >> ASTERISK is my gateway in My Asterisk is getting FreeSWITCH as CallerID >> not evean as CallerNAme. >> >> >> >> >> >> Thanks >> Jagan >> >> >> >> >> >> On 17 May 2013 13:01, Muhammad Shahzad wrote: >> >>> Well, if you want to have variable caller id number and name then use >>> "effective_caller_id_name" and "effective_caller_id_number" channel >>> variables in your dial plan. Beispiel, >>> >>> >>> >>> >>> OR if you want to use a global outbound caller id name and number then >>> you can modify following in vars.xml, >>> >>> >>> >>> >>> Thank you. >>> >>> >>> >>> >>> On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam < >>> jaganthoutam at gmail.com> wrote: >>> >>>> Any help on this issue. >>>> >>>> sent from samsung S3 >>>> On 17 May 2013 11:32, "Jagadish Thoutam" >>>> wrote: >>>> >>>>> No, it didn't help me Still i am getting FreeSWITH as a caller ID >>>>> (not Caller ID Name) caller id Name i am getting Number. >>>>> >>>>> >>>>> >>>>> Thanks >>>>> Jagan >>>>> >>>>> >>>>> On 16 May 2013 20:53, wrote: >>>>> >>>>>> Try on profile setting. >>>>>> >>>>>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>>>>> > HI, >>>>>> > >>>>>> > >>>>>> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >>>>>> > and Outbound calls with media proxy, here issue comes My Freeswith >>>>>> is >>>>>> > Sending FreeSwitch as a callid Number.. and >>>>>> > callid name as a Number i wondered why it is sending like that >>>>>> > i even comment the effective acller id name and caller id but still >>>>>> > same issue, can any one help on this. >>>>>> > >>>>>> > Version : >>>>>> > >>>>>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>>>>> > e1c325d >>>>>> > 2013-04-23 19:49:07Z) >>>>>> > >>>>>> > Thanks >>>>>> > Jagan >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Mit freundlichen Gr??en >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Mit freundlichen Gr??en > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/8631d042/attachment-0001.html From msc at freeswitch.org Sat May 18 03:29:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 17 May 2013 16:29:01 -0700 Subject: [Freeswitch-users] One Way Audio In-Reply-To: <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> Message-ID: It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy wrote: > Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed.*** > * > > ** ** > > My wife had just called my cell and got one way audio. So I ran: > pcapsipdump -f -p -i eth0 -d /tmp -n **** > > Of course I got 2 way audio. I called the one that ALWAYS fails ?. Got 2 > way audio! Does pcapsipdump fix it? lol**** > > ** ** > > I will try in day time tomorrow and see if we can get a failure.**** > > ** ** > > Nothing in the freeswitch.log of value? I didn?t see anything, but there > is still a lot for me to learn there.**** > > ** ** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, May 16, 2013 7:35 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] One Way Audio**** > > ** ** > > Sean,**** > > Glad to hear you're making progress with using tcpdump and other packet > capture-ish tools. You've successfully captured the SIP call leg between > your phone and your FreeSWITCH. That's good, but it's incomplete. You > really want SIP and RTP both, and you want the call leg between FreeSWITCH > and the telco. You have a few options:**** > > Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. > Con: creates massive pcap files through which you must sift to find the > call in question.**** > > Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for > each call leg into a single file. Cons: You have to compile it yourself, > and it creates a whole lot of files, so you'll need to get used to it.**** > > My personal opinion is this: if you never, ever have to debug a SIP call > ever again then just use tcpdump. However, if you're the phone guy and > you'll be doing this again in the future then bite the bullet and learn > pcapsipdump. Believe me it's worth it.**** > > -MC**** > > ** ** > > On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: > **** > > **** > > Hi all,**** > > **** > > First, I am on version: FreeSWITCH Version > 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) > I hope to move to the Stable 1.2.9 this weekend.**** > > **** > > I am having very reliable one way audio when calling Sprint Cell Phone > users, though not every time. I got this reproducible on my phone, but by > the time I learned tcpdump command, it was working again. However, the > user that reported it seems to get it almost everytime.**** > > **** > > Helpful tidbits:**** > > ? I THINK it happens in either direction.**** > > ? For this person at his home, it appears to be every time (for > now)**** > > ? He reports calling (to or from) other Sprint Cell users results > in the same problem from our FS**** > > ? It appears to only be true with Sprint Cell calls! (But my > users say that?s not Sprints fault!)**** > > **** > > Scenario:**** > > I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and > it rings, he answers, I can hear him crystal clear ? he can?t hear me at > all. I had a theory that it worked after 30 seconds (on my cell), but that > does not hold true on his cell.**** > > **** > > Here is the FS logfile: > http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** > ** > > And here is the tcpdump output: > http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip **** > > **** > > Based on the small size of the file, I suspect someone is going to say ?do > it again with this tcpdump command?. I welcome the education.**** > > **** > > Anyway, any insight will be appreciated.**** > > **** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/77b55707/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/77b55707/attachment.gif From philippe at ppmt.org Sat May 18 05:27:19 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Fri, 17 May 2013 21:27:19 -0400 Subject: [Freeswitch-users] advise to troubleshoot a call In-Reply-To: References: <172901ce4064$47b93370$d72b9a50$@bizfocused.com> <517735B0.6080502@ppmt.org> <5178815F.70806@ppmt.org> Message-ID: <5196D8F7.1090508@ppmt.org> I thought I would give an update on my audio problem. No ground breaking news and a bit out of topic, so if you don't like happy ending just ignore ! After restarting/resetting just about everything in my home (guruplug, router,cable box, the phone adapter) I run out of idea and was getting really desperate...my next step was to reboot the microwave oven just in case :) I then noticed while doing speed test on my line that the result were most of the time good but sometime would go very low then bounce back at the next test. I ended up calling my internet provider who after doing a few remote test told me that my connection was rather flaky and that he was going to send me a technician the next day. They ended replacing my cable box who had a bad battery but the technician also changed a few outdoor connectors that he said were about to give up and were most certainly the cause of my problem. Since the phone is working great but more important the wife is happy again! And I even get slightly more downlink/uplink that is actually advertised Thanks all for the help and especially Michel Brabants who really helped me understanding QOS. /Philippe On 13-04-25 01:32 PM, Philippe Le Toquin wrote: > Regarding my comment about not finding what has changed, I didn't mean > to imply it was something in Freeswitch. > > I am actually fairly certain it is not FS since it was working fine > for weeks on the same version. > > > > > On 25 April 2013 13:15, Michael Collins > wrote: > > > > > On Wed, Apr 24, 2013 at 6:05 PM, Philippe Le Toquin > > wrote: > > OK I installed pcapsipdump but it is killing my poor Guruplug. > > CPU is at 99% so the call are going through but there is a big > delay. So it is not going to help! > > Bummer! > > > What bugs is that up to 2/3 weeks ago everything was fine and > suddenly the cutout started. > I can't find out what can have change that would cause this. > > This is why I mentioned in another thread that I would like to get > people more familiar with git bisect. You can manually use git > bisect to find out which commit broke things for you. Check out > Mitch Capper's handy wiki entry: > http://wiki.freeswitch.org/wiki/Git_Tips#Git_Bisect_-_Tracking_down_breaks_and_bugs_extremely_fast_with_git > > -MC > > > /Philippe > > On 13-04-23 11:41 PM, Michael Collins wrote: >> pcapsipdump ftw! >> >> Check it out: >> http://wiki.freeswitch.org/wiki/Packet_Capture#pcapsipdump >> >> It gets pcaps for you and separates each call into its own >> file with both signaling and media. Find the pcap with a bad >> phone call and open that one up in Wireshark for analysis. >> >> -MC >> >> >> On Tue, Apr 23, 2013 at 6:30 PM, Philippe Le Toquin >> > wrote: >> >> Hi Sean and Michel >> >> Thanks for your input. >> >> I did try the test and they came back as all good saying >> I should get good quality (my link 10Mbps download, 1Mbps >> upload) >> The MOS score is reported as 4.2 >> >> What I didn't find is what you mention about 1, 3 or 5 >> lines. >> >> I will continue looking... >> >> >> On 13-04-23 05:45 PM, Michel Brabants wrote: >>> RTCP provides you normally with statistics that can help >>> you. I didn't check it in freeswitch yet, but there is >>> an option to enable it (not standard apparently?) in the >>> profile. One step further is rtcp-xr which can provide >>> mos-score, but with all of this, the receiving endpoint >>> should also support this. >>> >>> Kind regards, >>> >>> Michel >>> >>> On Tue, Apr 23, 2013 at 10:51 PM, Sean Devoy >>> > >>> wrote: >>> >>> Philippe, >>> >>> The audio is sent through UDP packets, not TCP >>> packets. The main difference is that if UDP packets >>> are lost in collisions or congestion, they are not >>> retransmitted. What you are describing COULD be >>> completely explained as internet congestion. >>> >>> I would test with some application that loads her >>> network link down. One example is >>> http://www.onsip.com/tools/voip-test >>> >>> Try the various levels (1 line, 3 or 5 lines) and >>> see how a call with her reacts when the test are >>> run. That will at a minimum rule out congestion on >>> her local internet connection. >>> >>> HTH, >>> >>> Sean >>> >>> *From:*freeswitch-users-bounces at lists.freeswitch.org >>> >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org ] >>> *On Behalf Of *Philippe Le Toquin >>> *Sent:* Tuesday, April 23, 2013 4:12 PM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] advise to troubleshoot >>> a call >>> >>> Hello, >>> >>> Recently I (well my wife) is having voice problem >>> when she is on a call. >>> >>> Typically she can hear people but they can't >>> >>> The problem is that it is not all the time and when >>> it happens it last a few seconds then all back to >>> normal... then later on it happens again. >>> >>> I am still in the process of trying to find if there >>> is a pattern in the problem but I would like to be >>> able to look at the log from FS. >>> >>> What would you recommend to be able to look at the >>> log (what debug level). >>> >>> Any tips to make it easier to identify an issue in >>> the log (like message to look for?) >>> >>> Thanks >>> >>> Philippe >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> >>> >>> 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 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130517/9b69fc77/attachment-0001.html From ashish at nms.co.in Sat May 18 14:08:16 2013 From: ashish at nms.co.in (Ashish gautam) Date: Sat, 18 May 2013 15:38:16 +0530 Subject: [Freeswitch-users] FreeTDM PRI channel monitoring Message-ID: Hi, I want to get the status of each PRI channel through event socket. I am sending api command "api ftdm dump " to get the status of each channel one by one. Its returning lots of other information also which I do not want. Is there any other efficient and recommended way to do this? Regards, -- Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/fbde683b/attachment.html From jaganthoutam at gmail.com Sat May 18 17:38:50 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Sat, 18 May 2013 09:38:50 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Hi Michael, i am not sure its strange because i tried everything what you said and even i am working with freeswitch from last 2 years i didn't have problem like this But still the same thing. callerid = FreeSWITCH caleerid name = +1XXXXXXXXXX Thanks Jagan On 17 May 2013 19:20, Michael Collins wrote: > > > > On Fri, May 17, 2013 at 3:24 PM, Muhammad Shahzad wrote: > >> How are you disabling the global caller id? This won't work, (its a >> common mistake done by many users when dealing with X-PRE-PROCESS) >> >> >> >> You have to do something like this, >> >> >> > FYI, we recommend not using XML comments when trying to disable a > preprocess directive. Just do something like this: > > > > That will do the trick and will make the code easier to read. > -MC > > >> >> Notice the space around ' - ' in X-PRE-PROCESS. >> >> Also you should set some appropriate caller id for outbound call in your >> dial plan, as already i told you in my previous email. >> >> Thank you. >> >> >> >> >> On Fri, May 17, 2013 at 10:26 PM, Jagadish Thoutam < >> jaganthoutam at gmail.com> wrote: >> >>> Hi Muhammad, >>> >>> i disable the global outbound caller id, >>> >>> My setup is like this >>> >>> DID -----------> FreeSWITCH -------> AsteriskServer >>> >>> >>> >>> >>> >> expression="^(?:\+1)(88XXXXXXXX)$"> >>> >>> >>> >>> >>> >>> ASTERISK is my gateway in My Asterisk is getting FreeSWITCH as >>> CallerID not evean as CallerNAme. >>> >>> >>> >>> >>> >>> Thanks >>> Jagan >>> >>> >>> >>> >>> >>> On 17 May 2013 13:01, Muhammad Shahzad wrote: >>> >>>> Well, if you want to have variable caller id number and name then use >>>> "effective_caller_id_name" and "effective_caller_id_number" channel >>>> variables in your dial plan. Beispiel, >>>> >>>> >>>> >>> data="effective_caller_id_number=49185551212"/> >>>> >>>> OR if you want to use a global outbound caller id name and number then >>>> you can modify following in vars.xml, >>>> >>>> >>>> >>>> >>>> Thank you. >>>> >>>> >>>> >>>> >>>> On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam < >>>> jaganthoutam at gmail.com> wrote: >>>> >>>>> Any help on this issue. >>>>> >>>>> sent from samsung S3 >>>>> On 17 May 2013 11:32, "Jagadish Thoutam" >>>>> wrote: >>>>> >>>>>> No, it didn't help me Still i am getting FreeSWITH as a caller ID >>>>>> (not Caller ID Name) caller id Name i am getting Number. >>>>>> >>>>>> >>>>>> >>>>>> Thanks >>>>>> Jagan >>>>>> >>>>>> >>>>>> On 16 May 2013 20:53, wrote: >>>>>> >>>>>>> Try on profile >>>>>>> setting. >>>>>>> >>>>>>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>>>>>> > HI, >>>>>>> > >>>>>>> > >>>>>>> > i have a freeswitch as a VOIP Gateway that will handle the Inbound >>>>>>> > and Outbound calls with media proxy, here issue comes My Freeswith >>>>>>> is >>>>>>> > Sending FreeSwitch as a callid Number.. and >>>>>>> > callid name as a Number i wondered why it is sending like that >>>>>>> > i even comment the effective acller id name and caller id but >>>>>>> still >>>>>>> > same issue, can any one help on this. >>>>>>> > >>>>>>> > Version : >>>>>>> > >>>>>>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>>>>>> > e1c325d >>>>>>> > 2013-04-23 19:49:07Z) >>>>>>> > >>>>>>> > Thanks >>>>>>> > Jagan >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.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 >>>>>>> >>>>>> >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Mit freundlichen Gr??en >>>> Muhammad Shahzad >>>> ----------------------------------- >>>> CISCO Rich Media Communication Specialist (CRMCS) >>>> CISCO Certified Network Associate (CCNA) >>>> Cell: +49 176 99 83 10 85 >>>> MSN: shari_786pk at hotmail.com >>>> Email: shaheryarkh at googlemail.com >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Mit freundlichen Gr??en >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +49 176 99 83 10 85 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/197fd8f1/attachment-0001.html From john at telecube.com.au Sat May 18 20:28:19 2013 From: john at telecube.com.au (Telecube - John) Date: Sun, 19 May 2013 02:28:19 +1000 Subject: [Freeswitch-users] Complex IVRs using data from MySQL Message-ID: <5197AC23.5020404@telecube.com.au> Ok, if someone could point me in the right direction I would be very grateful. I need to integrate database lookups into multi level IVRs to gather call routing info. The lookups are for variable multi digit keypress data as well as multiple sets of routing information. So far I can see my options are xml_curl or hand the call off to a lua script to handle the processing. I can catch the multi digit keypresses with regex and give that detail to a lua script, no problem. I'm hoping to be able to use the mod_ivr process as much as possible but I'm struggling to see how I can get data back into the call flow after dishing the duty off to lua. Can anyone point me down the right path please? Should I be happy to pass it off to lua and finish the lookups and routing in there? Should I try to keep as much as possible in the core ivr? Thanks.. - John From bdfoster at davri.com Sat May 18 20:49:59 2013 From: bdfoster at davri.com (Brian Foster) Date: Sat, 18 May 2013 12:49:59 -0400 Subject: [Freeswitch-users] Complex IVRs using data from MySQL In-Reply-To: <5197AC23.5020404@telecube.com.au> References: <5197AC23.5020404@telecube.com.au> Message-ID: Whether or not you hand it back to mod_ivr or not, to set a channel variable in Lua: session:setVariable("varname", "varvalue"); This has the same effect as using the set application in the dialplan: You can read the variable in Lua like this: varname = session:getVariable("varname"); -BDF On May 18, 2013 12:35 PM, "Telecube - John" wrote: > Ok, if someone could point me in the right direction I would be very > grateful. > > I need to integrate database lookups into multi level IVRs to gather > call routing info. > > The lookups are for variable multi digit keypress data as well as > multiple sets of routing information. > > So far I can see my options are xml_curl or hand the call off to a lua > script to handle the processing. > > I can catch the multi digit keypresses with regex and give that detail > to a lua script, no problem. > > I'm hoping to be able to use the mod_ivr process as much as possible but > I'm struggling to see how I can get data back into the call flow after > dishing the duty off to lua. > > Can anyone point me down the right path please? > > Should I be happy to pass it off to lua and finish the lookups and > routing in there? > > Should I try to keep as much as possible in the core ivr? > > Thanks.. > > - John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130518/e58d6b0f/attachment.html From drk at drkngs.net Sat May 18 22:10:04 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Sat, 18 May 2013 11:10:04 -0700 Subject: [Freeswitch-users] Complex IVRs using data from MySQL In-Reply-To: Message-ID: <20130518181004.3811e942@mail.tritonwest.net> Just from my experence, it's a lot easier to write the whole logic, for a complex IVR in something (lua, perl, python, C#, F#) or anything else that FS supports, rather then trying to cram it in to the simple IVR application. Remember you can do foreign DB access or talk to anything else (web service, raw http service, etc...) from any of the languages directly. I do almost all external interaction with C# modules, runnig inside of FS. Makes life real easy :) --Dave _____ From: Brian Foster [mailto:bdfoster at davri.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 18 May 2013 09:49:59 -0700 Subject: Re: [Freeswitch-users] Complex IVRs using data from MySQL Whether or not you hand it back to mod_ivr or not, to set a channel variable in Lua: session:setVariable("varname", "varvalue"); This has the same effect as using the set application in the dialplan: You can read the variable in Lua like this: varname = session:getVariable("varname"); -BDF On May 18, 2013 12:35 PM, "Telecube - John" wrote: Ok, if someone could point me in the right direction I would be very grateful. I need to integrate database lookups into multi level IVRs to gather call routing info. The lookups are for variable multi digit keypress data as well as multiple sets of routing information. So far I can see my options are xml_curl or hand the call off to a lua script to handle the processing. I can catch the multi digit keypresses with regex and give that detail to a lua script, no problem. I'm hoping to be able to use the mod_ivr process as much as possible but I'm struggling to see how I can get data back into the call flow after dishing the duty off to lua. Can anyone point me down the right path please? Should I be happy to pass it off to lua and finish the lookups and routing in there? Should I try to keep as much as possible in the core ivr? Thanks.. - John _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130518/fab6fd86/attachment.html From john at telecube.com.au Sat May 18 22:55:44 2013 From: john at telecube.com.au (Telecube - John) Date: Sun, 19 May 2013 04:55:44 +1000 Subject: [Freeswitch-users] Complex IVRs using data from MySQL In-Reply-To: <20130518181004.3811e942@mail.tritonwest.net> References: <20130518181004.3811e942@mail.tritonwest.net> Message-ID: <5197CEB0.1070804@telecube.com.au> Hi guys, Thanks for the info, I'm still trying to find the best solution for me. It's looking like handling it all in a lua script might be the way to go. However I would welcome expert opinions on the following concept. The call routes into the IVR and the following entry captures a 4 digit keypress and sends the data to the lua script: And either: The script does the db lookup and retrieves the destination number from the database and sends the call back into the default context: table.insert(ACTIONS, {"transfer", "0398776544 XML default"}) The destination number is matched and routed out as normal. or The script retrieves further detail from the database and sends the call back into the ivr_next_level context and we do the dance again: table.insert(ACTIONS, {"transfer", "101323556 XML ivr_next_level"}) In terms of speed and system resources is this an efficient way to handle a variable keypress IVR with possible multiple levels? - John On 19/05/13 4:10 AM, Dave R. Kompel wrote: > Just from my experence, it's a lot easier to write the whole logic, > for a complex IVR in something (lua, perl, python, C#, F#) or anything > else that FS supports, rather then trying to cram it in to the simple > IVR application. Remember you can do foreign DB access or talk to > anything else (web service, raw http service, etc...) from any of the > languages directly. > I do almost all external interaction with C# modules, runnig inside of > FS. Makes life real easy :) > --Dave > > ------------------------------------------------------------------------ > *From:* Brian Foster [mailto:bdfoster at davri.com] > *To:* FreeSWITCH Users Help > [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Sat, 18 May 2013 09:49:59 -0700 > *Subject:* Re: [Freeswitch-users] Complex IVRs using data from MySQL > > Whether or not you hand it back to mod_ivr or not, to set a > channel variable in Lua: > > session:setVariable("varname", "varvalue"); > > This has the same effect as using the set application in the dialplan: > > > > You can read the variable in Lua like this: > > varname = session:getVariable("varname"); > > -BDF > > On May 18, 2013 12:35 PM, "Telecube - John" > wrote: > > Ok, if someone could point me in the right direction I would > be very > grateful. > > I need to integrate database lookups into multi level IVRs to > gather > call routing info. > > The lookups are for variable multi digit keypress data as well as > multiple sets of routing information. > > So far I can see my options are xml_curl or hand the call off > to a lua > script to handle the processing. > > I can catch the multi digit keypresses with regex and give > that detail > to a lua script, no problem. > > I'm hoping to be able to use the mod_ivr process as much as > possible but > I'm struggling to see how I can get data back into the call > flow after > dishing the duty off to lua. > > Can anyone point me down the right path please? > > Should I be happy to pass it off to lua and finish the lookups and > routing in there? > > Should I try to keep as much as possible in the core ivr? > > Thanks.. > > - John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/56afba54/attachment-0001.html From sdevoy at bizfocused.com Sun May 19 00:59:44 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 18 May 2013 16:59:44 -0400 Subject: [Freeswitch-users] One Way Audio In-Reply-To: References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> Message-ID: <059e01ce540a$9f9a51a0$decef4e0$@bizfocused.com> Thanks MC. Let's shelve this for now. It appears to have been specific to one of our gateways. If it returns I will throw it at you. Thanks again. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/2ecc1acb/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/2ecc1acb/attachment.gif From sdevoy at bizfocused.com Sun May 19 01:22:03 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 18 May 2013 17:22:03 -0400 Subject: [Freeswitch-users] Detecting Errors when dialing through multiple gateways In-Reply-To: References: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> Message-ID: <05b301ce540d$bdfbb1e0$39f315a0$@bizfocused.com> Thanks Brian. I will dig in to some of the LUA samples. This is not the first time I have been steered that way. I am kind of glad to see you unavailable on Friday Night! Now we have to get MC a girlfriend or a favorite bar. He like answered EVERY open question at 7:30p on Friday night. (MAYBE he is in California and it was like 4:30p). Seriously though, thanks everyone for all the help. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: Friday, May 17, 2013 5:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Detecting Errors when dialing through multiple gateways Lua scripts are really good at doing this, and if you are crafty enough you can even give them a failure description and supporting sip trace. That IMO would be the best option. I'm mobile right now so I can't write you an example but maybe others can chime in. - BDF On May 17, 2013 4:59 PM, "Sean Devoy" wrote: Hi All, I have been having pretty good luck with redundancy by dialing through multiple gateways like: If gateway cheap fails, the call goes through on the expensive gateway. That is fine, but I would like an email notification of the failure on the cheap gateway. Is there any way to do that? The support group at the gateway vender is very good at correcting the problem when notified. I would love to send them an email of failures automatically. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130518/2242ead4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/2242ead4/attachment-0001.gif From sdevoy at bizfocused.com Sun May 19 01:22:21 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 18 May 2013 17:22:21 -0400 Subject: [Freeswitch-users] Detecting Errors when dialing through multiple gateways In-Reply-To: References: <00e701ce5340$a543f630$efcbe290$@bizfocused.com> Message-ID: <05b901ce540d$c8cf7430$5a6e5c90$@bizfocused.com> Awww, but there is a trade-off here. In my bridge, the "back up" expensive gateway starts connecting after only 8 seconds. Using your schema, in some cases, the first or "cheap" gateway takes the full timeout limit to expire. Only then, after 25 or 30 seconds, does the backup start trying to connect. Too many users give up after 20 or 25 seconds of silence. Few will wait for a 25 second timeout, followed by a 12 to 15 second connection. But, if error detection is not possible in the sequential dial way, I like your approach, but with an extra "play" message saying "ooops, rerouting, hang on, hang on I can do this" From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Detecting Errors when dialing through multiple gateways On Fri, May 17, 2013 at 1:53 PM, Sean Devoy wrote: Hi All, I have been having pretty good luck with redundancy by dialing through multiple gateways like: I'm curious why your're doing simultaneous dial instead of sequential dial here. In any case, you can have multiple bridge apps instead of doing the comma-sep list: Try it out. Also, don't forget the standard disclaimer . -MC If gateway cheap fails, the call goes through on the expensive gateway. That is fine, but I would like an email notification of the failure on the cheap gateway. Is there any way to do that? The support group at the gateway vender is very good at correcting the problem when notified. I would love to send them an email of failures automatically. Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/ab453370/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130518/ab453370/attachment.gif From fs-list at communicatefreely.net Sun May 19 08:06:29 2013 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Sun, 19 May 2013 00:06:29 -0400 Subject: [Freeswitch-users] Complex IVRs using data from MySQL In-Reply-To: <5197AC23.5020404@telecube.com.au> References: <5197AC23.5020404@telecube.com.au> Message-ID: <51984FC5.8000005@communicatefreely.net> Hi John, You can use xml_curl to build configuration for the simple IVR module on the fly. Every time the IVR application is called, there will be a configuration request, and in the request data is the name of the IVR that was called. You can run a server side script on your CURL server that will build the relevant IVR configuration and send it back. That's only one transaction, and it's pretty light weight. I'm not sure if it will work for your application or not, but I think it would. In our case, I have a table that is IVR menus (unique key, timeout, what sounds to play, etc.) I have another table that includes all the entries, and what action to take on a match. Any time an IVR is called, it can lookup the ivr row to create the IVR, and then iterate through all the entry rows to create all the options. If you have multi-level IVR trees, just make sure that you return all the IVRs that could be referenced through the sub menu action, so that they are available as well. There isn't a second config request when a sub-menu is run. Hope that helps! -Tim On 13-05-18 12:28 PM, Telecube - John wrote: > Ok, if someone could point me in the right direction I would be very > grateful. > > I need to integrate database lookups into multi level IVRs to gather > call routing info. > > The lookups are for variable multi digit keypress data as well as > multiple sets of routing information. > > So far I can see my options are xml_curl or hand the call off to a lua > script to handle the processing. > > I can catch the multi digit keypresses with regex and give that detail > to a lua script, no problem. > > I'm hoping to be able to use the mod_ivr process as much as possible but > I'm struggling to see how I can get data back into the call flow after > dishing the duty off to lua. > > Can anyone point me down the right path please? > > Should I be happy to pass it off to lua and finish the lookups and > routing in there? > > Should I try to keep as much as possible in the core ivr? > > Thanks.. > > - John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From cal.leeming at simplicitymedialtd.co.uk Sun May 19 08:14:55 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 19 May 2013 05:14:55 +0100 Subject: [Freeswitch-users] Bounty - FS-5446 Message-ID: Hello all, I have thrown up a bounty on the following ticket; http://jira.freeswitch.org/browse/FS-5446 Any takers? Thanks Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/d7d81fa8/attachment.html From intralanman at freeswitch.org Sun May 19 16:16:59 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Sun, 19 May 2013 08:16:59 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: I'm assuming we can look forward to seeing Francis at ClueCon this year too, then? Along with the rest of the 2600hz team ;-) -Ray On May 16, 2013 5:31 PM, "Joshua Goldbard" wrote: > Looping Francis in so you guys can yell at him belligerently :D. > > Cheers, > Joshua > > Joshua Goldbard > VP of Marketing, 2600hz > > 116 Natoma Street, Floor 2 > San Francisco, CA, 94104 > 415.886.7923 | j at 2600hz.com > > On May 16, 2013, at 2:18 PM, Darren Schreiber > wrote: > > Since Francis is now in charge of the project, I'll ask him to do it > :-) Then you guys will get to know him as well. He's very sharp and will > not really have anything but this project to work on for a while, and phone > provisioning. > > So, short answer, yes, I'll be happy to :-) > > - Darren > > From: Nick Giannak > Organization: HKC Radio > Reply-To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Date: Wednesday, May 15, 2013 5:23 PM > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX > > Darren, > Can you do us a favor and post here when it's fixed, so we can play > with it again? You once recommended Kazoo to me but it was so far beyond > the scope of what I needed or knew what to do with. It's nice to see > blue.box back in development! > Thanks, > Nick > On 5/15/2013 3:30 PM, Darren Schreiber wrote: > > Hi Mehroz, > There is an issue with the current git and nobody here has had the time to > fix it, but the ISO works. > > That said, the project has become dated, I agree. It's definitely not > dead though. We are actually rebuilding both blue.box items this week (the > ISO and fixing git), which should fix the build. It's definitely not dead. > > As of last week, Francis has been dedicated full-time on blue.box. You > will see activity on it via that route. > > You can also always ask questions in #2600hz. > > - Darren > > From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Date: Tuesday, May 14, 2013 11:44 PM > To: "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX > > About FusionPBX, i guess this is not a complete opensource solution, the > dialplan area is hidden and you cannot change it according to your > requirements (correct me if i am wrong)... overall functionality is good > and support is alive. > > and yes, bluebox seems to be dead now, howeverm very nice interface, > completely opensource. i am having some issues in its customization and > there seems to be no support alive! > > also note that, the iso image has different bluebox interface and git > version has different! and no one to help! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/b77a2721/attachment-0001.html From j at 2600hz.com Sun May 19 16:27:55 2013 From: j at 2600hz.com (Joshua Goldbard) Date: Sun, 19 May 2013 12:27:55 +0000 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: , Message-ID: We're planning on going to Hang out with those who are strong with the clue :). We <3 FreeSWITCH and want to give back to the community both via sponsoring the conference and participating in the event. The details are still being sorted and I'm confident we'll have more details soon :). I'll chat with Darren to see if we can make a Francis sighting happen. Just so I'm clear, you want Francis to come so you guys can yell at him until he cries right? :) just kidding! Francis is awesome (he's also the lead on our provisioning work too) and I'm sure he'd love to go if he has the time. But yes, the 2600hz folks are looking forward to cluecon, as we do every year. Cheers, Joshua Sent from my iPhone On May 19, 2013, at 5:21 AM, "Raymond Chandler" > wrote: I'm assuming we can look forward to seeing Francis at ClueCon this year too, then? Along with the rest of the 2600hz team ;-) -Ray On May 16, 2013 5:31 PM, "Joshua Goldbard" > wrote: Looping Francis in so you guys can yell at him belligerently :D. Cheers, Joshua Joshua Goldbard VP of Marketing, 2600hz 116 Natoma Street, Floor 2 San Francisco, CA, 94104 415.886.7923 | j at 2600hz.com On May 16, 2013, at 2:18 PM, Darren Schreiber > wrote: Since Francis is now in charge of the project, I'll ask him to do it :-) Then you guys will get to know him as well. He's very sharp and will not really have anything but this project to work on for a while, and phone provisioning. So, short answer, yes, I'll be happy to :-) - Darren From: Nick Giannak > Organization: HKC Radio Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Wednesday, May 15, 2013 5:23 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX Darren, Can you do us a favor and post here when it's fixed, so we can play with it again? You once recommended Kazoo to me but it was so far beyond the scope of what I needed or knew what to do with. It's nice to see blue.box back in development! Thanks, Nick On 5/15/2013 3:30 PM, Darren Schreiber wrote: Hi Mehroz, There is an issue with the current git and nobody here has had the time to fix it, but the ISO works. That said, the project has become dated, I agree. It's definitely not dead though. We are actually rebuilding both blue.box items this week (the ISO and fixing git), which should fix the build. It's definitely not dead. As of last week, Francis has been dedicated full-time on blue.box. You will see activity on it via that route. You can also always ask questions in #2600hz. - Darren From: Mehroz Ashraf > Reply-To: "freeswitch-users at lists.freeswitch.org" > Date: Tuesday, May 14, 2013 11:44 PM To: "freeswitch-users at lists.freeswitch.org" > Subject: Re: [Freeswitch-users] Recommendation for GUI to PBX About FusionPBX, i guess this is not a complete opensource solution, the dialplan area is hidden and you cannot change it according to your requirements (correct me if i am wrong)... overall functionality is good and support is alive. and yes, bluebox seems to be dead now, howeverm very nice interface, completely opensource. i am having some issues in its customization and there seems to be no support alive! also note that, the iso image has different bluebox interface and git version has different! and no one to help! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-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://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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130519/1cb4483a/attachment.html From intralanman at freeswitch.org Sun May 19 16:44:34 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Sun, 19 May 2013 08:44:34 -0400 Subject: [Freeswitch-users] Recommendation for GUI to PBX In-Reply-To: References: Message-ID: > I'll chat with Darren to see if we can make a Francis sighting happen. Just so I'm clear, you want Francis to come so you guys can yell at him until he cries right? :) I'll tell him a few stories about things I've run into while working on provisioning for Cudatel, that should make him cry. :-D -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/620cd876/attachment.html From john at telecube.com.au Sun May 19 17:57:57 2013 From: john at telecube.com.au (Telecube - John) Date: Sun, 19 May 2013 23:57:57 +1000 Subject: [Freeswitch-users] UUID - how unique is it? Message-ID: <5198DA65.20201@telecube.com.au> How unique is uuid in the cdr records? Coming from asterisk where uid is about as unique as a black bowling ball.... Can I count on uuid to be totally unique in my billing records? -- Regards, John Matich Telecube Pty Ltd www.telecube.com.au Ph: 13CUBE (132823) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/bda5abca/attachment-0001.html From nneul at mst.edu Sun May 19 18:17:58 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Sun, 19 May 2013 09:17:58 -0500 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data Message-ID: <5198DF16.5000201@mst.edu> I've noticed several places in FS code and examples where it isn't safe at all to take user supplied data. An easy example is the use of mailer_app: #ifdef WIN32 switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", runtime.mailer_app, from, runtime.mailer_app_args, to, filename); #else switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", filename, runtime.mailer_app, from, runtime.mailer_app_args, to); #endif another is ANY use of passing channel vars or data to a system or bgsystem command. This isn't an issue normally, but if you want to give limited ability for users to control their own dial rules, then you wind up having to be very careful with processing the data to make sure it's safe. That's always a good idea, but it still seems like a bad idea to take that data and then directly use it in a completely unsafe context like a parsed command line. For the voicemail notify case, seems like an easy answer would be something like a "vm-notify-hook", which at that point, could call out to lua or perl to do the actual sending in a safe manner, passing the recipient/sender/etc. as data instead of on cmd line. For the 'passing channel vars...' case, I think it would be good to have a 'system_json' and 'bgsystem_json' set of routines that would pass channel data to the script on stdin in json format. Regardless of implementation of either of those, I think it would be worthwhile to have a shell_escape() routine in the core utilities to allow the current syntax to be used more safely. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From ehermouet at bluetel.fr Sun May 19 18:29:01 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Sun, 19 May 2013 16:29:01 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> Message-ID: <446b2500-7975-4d51-87f0-90edee2d9a31@email.android.com> Nobody know ? Tks Michael Collins a ?crit?: >On Thu, May 16, 2013 at 2:38 AM, wrote: > >> hi >> >> i have this on my log >> >> 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2931 Set 2833 dtmf >> receive payload to 101 >> 2013-05-16 11:37:16.028265 [DEBUG] sofia_glue.c:2926 Set 2833 dtmf >send >> payload to 101 >> >> i'm sure i don't have configure my xml file but where and how ? >> yes provider use 2833 >> > >Can you pastebin the entire call from start to finish? Use >pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >highlighting. Reply to this email thread with the URL from pastebin. >-MC > > >> >> tks >> >> >> Le 2013-05-15 23:42, Michael Collins a ?crit : >> > I recommend that you get a console debug log of the call and put it >> > up >> > on pastebin.freeswitch.org [11]. Hopefully well be able to see what >> > is >> > happening. >> > >> > In the meantime, is your provider set to use RFC2833 digits or do >> > they want to see in-band DTMFs? >> > -MC >> > >> > On Wed, May 15, 2013 at 2:01 PM, Erwan Hermouet >> > [12]> wrote: >> > >> >> Hi all. We have voip provider connectd to FS and 1 extension. When >> >> my extension make call using voip provider dtmf not work. Hi phone >> >> to voip provider and dtmf is active to rfc2833. >> >> >> >> >> >> >> >> So i suppose i miss something on my config. Can you explain where >? >> >> tks advance >> >> >> >> >> >> >> > >> > >_________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [1] >> >> http://www.freeswitchsolutions.com [2] >> >> >> >> >> >> [3] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [4] >> >> http://wiki.freeswitch.org [5] >> >> http://www.cluecon.com [6] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [7] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8] >> >> >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [9] >> >> http://www.freeswitch.org [10] >> > >> > -- >> > Michael S Collins >> > Twitter: @mercutioviz >> > http://www.FreeSWITCH.org [13] >> > http://www.ClueCon.com [14] >> > http://www.OSTAG.org [15] >> > >> > >> > >> > Links: >> > ------ >> > [1] mailto:consulting at freeswitch.org >> > [2] http://www.freeswitchsolutions.com >> > [3] >> > [4] http://www.freeswitch.org >> > [5] http://wiki.freeswitch.org >> > [6] http://www.cluecon.com >> > [7] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [9] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [10] http://www.freeswitch.org >> > [11] http://pastebin.freeswitch.org >> > [12] mailto:ehermouet at bluetel.fr >> > [13] http://www.FreeSWITCH.org >> > [14] http://www.ClueCon.com >> > [15] http://www.OSTAG.org >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > >-- >Michael S Collins >Twitter: @mercutioviz >http://www.FreeSWITCH.org >http://www.ClueCon.com >http://www.OSTAG.org > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/c4dab877/attachment.html From jleung at v10networks.ca Sun May 19 19:06:51 2013 From: jleung at v10networks.ca (Jeff Leung) Date: Sun, 19 May 2013 08:06:51 -0700 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <5198DA65.20201@telecube.com.au> References: <5198DA65.20201@telecube.com.au> Message-ID: <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> It depends on what implementation of the UUID library the build system has detected. If you've had the libuuid from e2fsprogs (Linux at the very least) installed and headers present, the build system will link against that. That implementation generates random UUID's for the most part. If you didn't have libuuid and its headers installed, the build system will use APR's own implementation for generating the UUID's; those generated UUID's are known to be sequential in nature. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Telecube - John Sent: Sunday, May 19, 2013 6:58 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] UUID - how unique is it? How unique is uuid in the cdr records? Coming from asterisk where uid is about as unique as a black bowling ball.... Can I count on uuid to be totally unique in my billing records? -- Regards, John Matich Telecube Pty Ltd www.telecube.com.au Ph: 13CUBE (132823) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/a0ddf6b5/attachment.html From sertys at gmail.com Sun May 19 23:21:59 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 19 May 2013 21:21:59 +0200 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data In-Reply-To: <5198DF16.5000201@mst.edu> References: <5198DF16.5000201@mst.edu> Message-ID: I am glad to see someone is concerned about input validation when it comes to voip. It is much neglected when we're constructing our services, partly due to the fact that it's still considered black magic. I believe that system and bgsystem should be strictly regulated and ani and sip vars should be safe-parsed before feeding to a turing machine. Security through obscurity has never worked and i beth my both legs we all have a few vulnerable applications behind our backs. Let's unite to make FS the most stable and secure softswitch out there. On May 19, 2013 5:21 PM, "Nathan Neulinger" wrote: > I've noticed several places in FS code and examples where it isn't safe at > all to take user supplied data. > > An easy example is the use of mailer_app: > > > #ifdef WIN32 > switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", > runtime.mailer_app, from, > runtime.mailer_app_args, to, filename); > #else > switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", > filename, runtime.mailer_app, from, > runtime.mailer_app_args, to); > #endif > > another is ANY use of passing channel vars or data to a system or bgsystem > command. > > > This isn't an issue normally, but if you want to give limited ability for > users to control their own dial rules, then > you wind up having to be very careful with processing the data to make > sure it's safe. That's always a good idea, but it > still seems like a bad idea to take that data and then directly use it in > a completely unsafe context like a parsed > command line. > > For the voicemail notify case, seems like an easy answer would be > something like a "vm-notify-hook", which at that > point, could call out to lua or perl to do the actual sending in a safe > manner, passing the recipient/sender/etc. as > data instead of on cmd line. > > For the 'passing channel vars...' case, I think it would be good to have a > 'system_json' and 'bgsystem_json' set of > routines that would pass channel data to the script on stdin in json > format. > > Regardless of implementation of either of those, I think it would be > worthwhile to have a shell_escape() routine in the > core utilities to allow the current syntax to be used more safely. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/e074e93d/attachment-0001.html From cal.leeming at simplicitymedialtd.co.uk Sun May 19 23:50:23 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Sun, 19 May 2013 20:50:23 +0100 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data In-Reply-To: References: <5198DF16.5000201@mst.edu> Message-ID: This could really use some input from Tony, as it really comes down to a design decision. It could be argued that you should not be exposing raw configuration to your customers without sanitizing the input yourselves.. and it could be argued that the input should be sanitized anyway. FS isn't attempting to push security through obscurity (given that all the code is open source), but many closed source products do. Either way, it would be good to hear Tony's thoughts on this. Cal On Sun, May 19, 2013 at 8:21 PM, Daniel Ivanov wrote: > I am glad to see someone is concerned about input validation when it comes > to voip. It is much neglected when we're constructing our services, partly > due to the fact that it's still considered black magic. I believe that > system and bgsystem should be strictly regulated and ani and sip vars > should be safe-parsed before feeding to a turing machine. Security through > obscurity has never worked and i beth my both legs we all have a few > vulnerable applications behind our backs. Let's unite to make FS the most > stable and secure softswitch out there. > On May 19, 2013 5:21 PM, "Nathan Neulinger" wrote: > >> I've noticed several places in FS code and examples where it isn't safe >> at all to take user supplied data. >> >> An easy example is the use of mailer_app: >> >> >> #ifdef WIN32 >> switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", >> runtime.mailer_app, from, >> runtime.mailer_app_args, to, filename); >> #else >> switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", >> filename, runtime.mailer_app, from, >> runtime.mailer_app_args, to); >> #endif >> >> another is ANY use of passing channel vars or data to a system or >> bgsystem command. >> >> >> This isn't an issue normally, but if you want to give limited ability for >> users to control their own dial rules, then >> you wind up having to be very careful with processing the data to make >> sure it's safe. That's always a good idea, but it >> still seems like a bad idea to take that data and then directly use it in >> a completely unsafe context like a parsed >> command line. >> >> For the voicemail notify case, seems like an easy answer would be >> something like a "vm-notify-hook", which at that >> point, could call out to lua or perl to do the actual sending in a safe >> manner, passing the recipient/sender/etc. as >> data instead of on cmd line. >> >> For the 'passing channel vars...' case, I think it would be good to have >> a 'system_json' and 'bgsystem_json' set of >> routines that would pass channel data to the script on stdin in json >> format. >> >> Regardless of implementation of either of those, I think it would be >> worthwhile to have a shell_escape() routine in the >> core utilities to allow the current syntax to be used more safely. >> >> -- Nathan >> >> ------------------------------------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/f15b3dea/attachment.html From nneul at mst.edu Mon May 20 00:26:25 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Sun, 19 May 2013 15:26:25 -0500 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data In-Reply-To: References: <5198DF16.5000201@mst.edu> Message-ID: <51993571.8040804@mst.edu> Agreed. I was actually (in my case) less concerned about security than 'oops that didn't do the right thing' for stuff as simple as "user email address(es)" - accidentally using a ";" to separate them (users coming from outlook for example) would result in a completely improper behavior - trying to run user address as a command... Similar for something like "send this caller-id name to this external system command" - too many places where a seemingly legitimate use is really insecure. I submitted a Jira with a patch to at least add a shell quoting function, and another one to make use of that shell quoting for the email addr in switch_simple_email. I'm sure there are others, but I wanted to cover a couple obvious cases. Would need more work to be fully useful though... -- Nathan On 05/19/2013 02:50 PM, Cal Leeming [Simplicity Media Ltd] wrote: > This could really use some input from Tony, as it really comes down to a design decision. > > It could be argued that you should not be exposing raw configuration to your customers without sanitizing the input > yourselves.. and it could be argued that the input should be sanitized anyway. > > FS isn't attempting to push security through obscurity (given that all the code is open source), but many closed source > products do. > > Either way, it would be good to hear Tony's thoughts on this. > > Cal > > On Sun, May 19, 2013 at 8:21 PM, Daniel Ivanov > wrote: > > I am glad to see someone is concerned about input validation when it comes to voip. It is much neglected when we're > constructing our services, partly due to the fact that it's still considered black magic. I believe that system and > bgsystem should be strictly regulated and ani and sip vars should be safe-parsed before feeding to a turing > machine. Security through obscurity has never worked and i beth my both legs we all have a few vulnerable > applications behind our backs. Let's unite to make FS the most stable and secure softswitch out there. > > On May 19, 2013 5:21 PM, "Nathan Neulinger" > wrote: > > I've noticed several places in FS code and examples where it isn't safe at all to take user supplied data. > > An easy example is the use of mailer_app: > > > #ifdef WIN32 > switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", runtime.mailer_app, from, > runtime.mailer_app_args, to, filename); > #else > switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", filename, runtime.mailer_app, from, > runtime.mailer_app_args, to); > #endif > > another is ANY use of passing channel vars or data to a system or bgsystem command. > > > This isn't an issue normally, but if you want to give limited ability for users to control their own dial rules, > then > you wind up having to be very careful with processing the data to make sure it's safe. That's always a good > idea, but it > still seems like a bad idea to take that data and then directly use it in a completely unsafe context like a parsed > command line. > > For the voicemail notify case, seems like an easy answer would be something like a "vm-notify-hook", which at that > point, could call out to lua or perl to do the actual sending in a safe manner, passing the recipient/sender/etc. as > data instead of on cmd line. > > For the 'passing channel vars...' case, I think it would be good to have a 'system_json' and 'bgsystem_json' set of > routines that would pass channel data to the script on stdin in json format. > > Regardless of implementation of either of those, I think it would be worthwhile to have a shell_escape() routine > in the > core utilities to allow the current syntax to be used more safely. > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From avi at avimarcus.net Mon May 20 01:06:12 2013 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 20 May 2013 00:06:12 +0300 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data In-Reply-To: <51993571.8040804@mst.edu> References: <5198DF16.5000201@mst.edu> <51993571.8040804@mst.edu> Message-ID: All of FS gives you unrestricted access -- raw XML, mod_xml_curl, lua, etc. If you want to limit access, use http://wiki.freeswitch.org/wiki/Mod_httapi -- it has a permissions option for what the user is allowed to access. THAT system has a permission framework. If you find a bug in there, then there's something to do about it. -Avi On Sun, May 19, 2013 at 11:26 PM, Nathan Neulinger wrote: > Agreed. I was actually (in my case) less concerned about security than > 'oops that didn't do the right thing' for stuff > as simple as "user email address(es)" - accidentally using a ";" to > separate them (users coming from outlook for > example) would result in a completely improper behavior - trying to run > user address as a command... > > Similar for something like "send this caller-id name to this external > system command" - too many places where a > seemingly legitimate use is really insecure. > > I submitted a Jira with a patch to at least add a shell quoting function, > and another one to make use of that shell > quoting for the email addr in switch_simple_email. I'm sure there are > others, but I wanted to cover a couple obvious cases. > > Would need more work to be fully useful though... > > -- Nathan > > On 05/19/2013 02:50 PM, Cal Leeming [Simplicity Media Ltd] wrote: > > This could really use some input from Tony, as it really comes down to a > design decision. > > > > It could be argued that you should not be exposing raw configuration to > your customers without sanitizing the input > > yourselves.. and it could be argued that the input should be sanitized > anyway. > > > > FS isn't attempting to push security through obscurity (given that all > the code is open source), but many closed source > > products do. > > > > Either way, it would be good to hear Tony's thoughts on this. > > > > Cal > > > > On Sun, May 19, 2013 at 8:21 PM, Daniel Ivanov sertys at gmail.com>> wrote: > > > > I am glad to see someone is concerned about input validation when it > comes to voip. It is much neglected when we're > > constructing our services, partly due to the fact that it's still > considered black magic. I believe that system and > > bgsystem should be strictly regulated and ani and sip vars should > be safe-parsed before feeding to a turing > > machine. Security through obscurity has never worked and i beth my > both legs we all have a few vulnerable > > applications behind our backs. Let's unite to make FS the most > stable and secure softswitch out there. > > > > On May 19, 2013 5:21 PM, "Nathan Neulinger" nneul at mst.edu>> wrote: > > > > I've noticed several places in FS code and examples where it > isn't safe at all to take user supplied data. > > > > An easy example is the use of mailer_app: > > > > > > #ifdef WIN32 > > switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < > \"%s\"\"", runtime.mailer_app, from, > > runtime.mailer_app_args, to, filename); > > #else > > switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s > %s %s", filename, runtime.mailer_app, from, > > runtime.mailer_app_args, to); > > #endif > > > > another is ANY use of passing channel vars or data to a system > or bgsystem command. > > > > > > This isn't an issue normally, but if you want to give limited > ability for users to control their own dial rules, > > then > > you wind up having to be very careful with processing the data > to make sure it's safe. That's always a good > > idea, but it > > still seems like a bad idea to take that data and then directly > use it in a completely unsafe context like a parsed > > command line. > > > > For the voicemail notify case, seems like an easy answer would > be something like a "vm-notify-hook", which at that > > point, could call out to lua or perl to do the actual sending in > a safe manner, passing the recipient/sender/etc. as > > data instead of on cmd line. > > > > For the 'passing channel vars...' case, I think it would be good > to have a 'system_json' and 'bgsystem_json' set of > > routines that would pass channel data to the script on stdin in > json format. > > > > Regardless of implementation of either of those, I think it > would be worthwhile to have a shell_escape() routine > > in the > > core utilities to allow the current syntax to be used more > safely. > > > > -- Nathan > > > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 FreeSWITCH-users at 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://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/487d198b/attachment-0001.html From steveayre at gmail.com Mon May 20 01:30:45 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 19 May 2013 22:30:45 +0100 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> Message-ID: libuuid generates one of several types of uuid. One simply makes all bits random from /dev/urandom. Collisions are unlikely but still possible. The time one allocates some bits to the mac and others to the unix timestamp, then the rest is sequential. That one means different servers will never generate collisions (assuming all use the same scheme), and collisions can't happen within the program on the server. That can be made between multiple programs using the library too. http://linux.die.net/man/3/uuid_generate Personally I would prefer the later but for privacy reasons (the mac) the first is the libraries default. Not sure which FS uses... On Sunday, May 19, 2013, Jeff Leung wrote: > It depends on what implementation of the UUID library the build system has > detected. > > If you?ve had the libuuid from e2fsprogs (Linux at the very least) > installed and headers present, the build system will link against that. > That implementation generates random UUID?s for the most part. If you > didn?t have libuuid and its headers installed, the build system will use > APR?s own implementation for generating the UUID?s; those generated UUID?s > are known to be sequential in nature.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org 'cvml', 'freeswitch-users-bounces at lists.freeswitch.org');> [mailto: > freeswitch-users-bounces at lists.freeswitch.org 'freeswitch-users-bounces at lists.freeswitch.org');>] *On Behalf Of *Telecube > - John > *Sent:* Sunday, May 19, 2013 6:58 AM > *To:* freeswitch-users at lists.freeswitch.org 'freeswitch-users at lists.freeswitch.org');> > *Subject:* [Freeswitch-users] UUID - how unique is it?**** > > ** ** > > How unique is uuid in the cdr records? > > Coming from asterisk where uid is about as unique as a black bowling > ball.... > > Can I count on uuid to be totally unique in my billing records?**** > > -- > Regards, > John Matich > Telecube Pty Ltd > www.telecube.com.au > Ph: 13CUBE (132823)**** > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/3e6c60bc/attachment.html From tatianava at ukr.net Sun May 19 13:48:12 2013 From: tatianava at ukr.net (Vitali) Date: Sun, 19 May 2013 02:48:12 -0700 (PDT) Subject: [Freeswitch-users] mod_spy problem Message-ID: <1368956892507-7590846.post@n2.nabble.com> Hi, I have a problem with mod_spy - it just plays moh and do not actually spy. Here`s my dialplan: / / When I dial 891000, for example, it`s just plays moh despite the fact that I`m making and receiving calls from user 1000. When I run a command /userspy_show/ in fs_cli, I get the following output: /: 54995932-2a7c-11de-af08-93e49196b898 1 total spy/ with no user supplied for spying. There`s no errors in logs. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spy-problem-tp7590846.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at hendrienet.com Sun May 19 17:25:17 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Sun, 19 May 2013 08:25:17 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID Message-ID: *==========================* *Server:* Ubuntu 12.04 *Version:* FreeSWITCH Version 1.5.1 *Enabled:* Multi-Tenant *Extensions:* 1000 and 1001 *Connection:* freeswitch server NAT to provider *Connection: *extensions are on the same LAN as fs server *DID:* 1 DID ========================== *Can-Do: * - I can dial extension to extension (1000 to 1001) and the phone rings and can talk. - I can test with extension 5000 and listen to the echo test. - I can dial the DID to the Freeswitch server. The server sends the the call to voicemail, 1000, when it should ring the phone. *Cannot Do:* - I cannot call out of any extension. The phone does not ring. It is silent. - I cannot dial the DID and ring a phone. It only goes to VoiceMail. *Question:* - Why can I not dial the DID and have the extension ring? - Where should I check? Thank you, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/5f3dc978/attachment.html From mishehu at freeswitch.org Mon May 20 02:41:04 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Sun, 19 May 2013 17:41:04 -0500 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> Message-ID: <51995500.2030504@freeswitch.org> If you're running multiple FS instances, you could also use both the FS server UUID in addition to the session UUID together. That'd mean storing more data but it would probably be on the order of winning a $500 million USD Powerball game twice within the same month... -Yossi From ben at langfeld.co.uk Mon May 20 03:10:04 2013 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sun, 19 May 2013 20:10:04 -0300 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <51995500.2030504@freeswitch.org> References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> <51995500.2030504@freeswitch.org> Message-ID: And if you can make that repeatable using libuuid, please do let me know ;) Regards, Ben Langfeld On 19 May 2013 19:41, Yossi Neiman wrote: > If you're running multiple FS instances, you could also use both the FS > server UUID in addition to the session UUID together. That'd mean > storing more data but it would probably be on the order of winning a > $500 million USD Powerball game twice within the same month... > > -Yossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/55747d56/attachment.html From jaybinks at gmail.com Mon May 20 03:53:32 2013 From: jaybinks at gmail.com (jay binks) Date: Mon, 20 May 2013 09:53:32 +1000 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <5198DA65.20201@telecube.com.au> References: <5198DA65.20201@telecube.com.au> Message-ID: In my case ( where there are multiple servers ) I pre-ended the server ID, before the UUID just to reduce the possibility of collisions. I guess it just depends how many calls you figure your going to have on your platform. On 19 May 2013 23:57, Telecube - John wrote: > How unique is uuid in the cdr records? > > Coming from asterisk where uid is about as unique as a black bowling > ball.... > > Can I count on uuid to be totally unique in my billing records? > -- > Regards, > John Matich > Telecube Pty Ltd > www.telecube.com.au > Ph: 13CUBE (132823) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/eda7a864/attachment-0001.html From john at telecube.com.au Mon May 20 04:21:42 2013 From: john at telecube.com.au (Telecube - John) Date: Mon, 20 May 2013 10:21:42 +1000 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: References: <5198DA65.20201@telecube.com.au> Message-ID: <51996C96.3030205@telecube.com.au> Hi guys, Thanks for the responses, I think I'll stick with the current process of writing a db entry for each call that comes in and using that insert id as the call event id. That guarantees uniqueness but requires an extra trip to the db to get it. Cheers, John On 20/05/13 9:53 AM, jay binks wrote: > In my case ( where there are multiple servers ) I pre-ended the server > ID, before the UUID > just to reduce the possibility of collisions. > > I guess it just depends how many calls you figure your going to have > on your platform. > > > > On 19 May 2013 23:57, Telecube - John > wrote: > > How unique is uuid in the cdr records? > > Coming from asterisk where uid is about as unique as a black > bowling ball.... > > Can I count on uuid to be totally unique in my billing records? > -- > Regards, > John Matich > Telecube Pty Ltd > www.telecube.com.au > Ph: 13CUBE (132823 ) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Sincerely > > Jay > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/050137f9/attachment.html From nneul at mst.edu Mon May 20 05:36:10 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Sun, 19 May 2013 20:36:10 -0500 Subject: [Freeswitch-users] Thoughts on security/code injection/etc. in FS when allowing user supplied data In-Reply-To: References: <5198DF16.5000201@mst.edu> <51993571.8040804@mst.edu> Message-ID: <51997E0A.8010800@mst.edu> I'm not giving users direct access to the system/APIs/etc - it's just via a portal/UI - but there's a difference between "worry about user screwing up their dialplan from me missing something if I let them define an email address" vs "give user full unrestricted access cause the use of email address in the core doesn't even quote command line arguments". In both cases, I still need to be concerned with sanitizing input - more a matter of what the defense-in-depth looks like. Right now - if I _want_ to allow untrusted input in certain circumstances - such as passing caller id info to an external executable, there simply is no way to do it safely. That's what the patches are for. -- Nathan On 05/19/2013 04:06 PM, Avi Marcus wrote: > All of FS gives you unrestricted access -- raw XML, mod_xml_curl, lua, etc. > > If you want to limit access, use http://wiki.freeswitch.org/wiki/Mod_httapi -- it has a permissions option for what the > user is allowed to access. > THAT system has a permission framework. If you find a bug in there, then there's something to do about it. > -Avi > > On Sun, May 19, 2013 at 11:26 PM, Nathan Neulinger > wrote: > > Agreed. I was actually (in my case) less concerned about security than 'oops that didn't do the right thing' for stuff > as simple as "user email address(es)" - accidentally using a ";" to separate them (users coming from outlook for > example) would result in a completely improper behavior - trying to run user address as a command... > > Similar for something like "send this caller-id name to this external system command" - too many places where a > seemingly legitimate use is really insecure. > > I submitted a Jira with a patch to at least add a shell quoting function, and another one to make use of that shell > quoting for the email addr in switch_simple_email. I'm sure there are others, but I wanted to cover a couple obvious > cases. > > Would need more work to be fully useful though... > > -- Nathan > > On 05/19/2013 02:50 PM, Cal Leeming [Simplicity Media Ltd] wrote: > > This could really use some input from Tony, as it really comes down to a design decision. > > > > It could be argued that you should not be exposing raw configuration to your customers without sanitizing the input > > yourselves.. and it could be argued that the input should be sanitized anyway. > > > > FS isn't attempting to push security through obscurity (given that all the code is open source), but many closed > source > > products do. > > > > Either way, it would be good to hear Tony's thoughts on this. > > > > Cal > > > > On Sun, May 19, 2013 at 8:21 PM, Daniel Ivanov > >> wrote: > > > > I am glad to see someone is concerned about input validation when it comes to voip. It is much neglected when > we're > > constructing our services, partly due to the fact that it's still considered black magic. I believe that > system and > > bgsystem should be strictly regulated and ani and sip vars should be safe-parsed before feeding to a turing > > machine. Security through obscurity has never worked and i beth my both legs we all have a few vulnerable > > applications behind our backs. Let's unite to make FS the most stable and secure softswitch out there. > > > > On May 19, 2013 5:21 PM, "Nathan Neulinger" >> wrote: > > > > I've noticed several places in FS code and examples where it isn't safe at all to take user supplied data. > > > > An easy example is the use of mailer_app: > > > > > > #ifdef WIN32 > > switch_snprintf(buf, B64BUFFLEN, "\"\"%s\" -f %s %s %s < \"%s\"\"", runtime.mailer_app, from, > > runtime.mailer_app_args, to, filename); > > #else > > switch_snprintf(buf, B64BUFFLEN, "/bin/cat %s | %s -f %s %s %s", filename, runtime.mailer_app, from, > > runtime.mailer_app_args, to); > > #endif > > > > another is ANY use of passing channel vars or data to a system or bgsystem command. > > > > > > This isn't an issue normally, but if you want to give limited ability for users to control their own dial > rules, > > then > > you wind up having to be very careful with processing the data to make sure it's safe. That's always a good > > idea, but it > > still seems like a bad idea to take that data and then directly use it in a completely unsafe context > like a parsed > > command line. > > > > For the voicemail notify case, seems like an easy answer would be something like a "vm-notify-hook", > which at that > > point, could call out to lua or perl to do the actual sending in a safe manner, passing the > recipient/sender/etc. as > > data instead of on cmd line. > > > > For the 'passing channel vars...' case, I think it would be good to have a 'system_json' and > 'bgsystem_json' set of > > routines that would pass channel data to the script on stdin in json format. > > > > Regardless of implementation of either of those, I think it would be worthwhile to have a shell_escape() > routine > > in the > > core utilities to allow the current syntax to be used more safely. > > > > -- Nathan > > > > ------------------------------------------------------------ > > Nathan Neulinger nneul at mst.edu > > > Missouri S&T Information Technology (573) 612-1412 > > System Administrator - Architect > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.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 > > > > > > -- > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From philippe at ppmt.org Mon May 20 05:55:06 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Sun, 19 May 2013 21:55:06 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: Message-ID: <5199827A.5080003@ppmt.org> Without seeing your dialplan, it is going difficult to help what does the command below gives you: sofia status On 13-05-19 09:25 AM, Mike Hendrie wrote: > *==========================* > *Server:* Ubuntu 12.04 > *Version:* FreeSWITCH Version 1.5.1 > *Enabled:* Multi-Tenant > *Extensions:* 1000 and 1001 > *Connection:* freeswitch server NAT to provider > *Connection: *extensions are on the same LAN as fs server > *DID:* 1 DID > ========================== > > *Can-Do: * > > * I can dial extension to extension (1000 to 1001) and the phone > rings and can talk. > * I can test with extension 5000 and listen to the echo test. > * I can dial the DID to the Freeswitch server. The server sends the > the call to voicemail, 1000, when it should ring the phone. > > *Cannot Do:* > > * I cannot call out of any extension. The phone does not ring. It is > silent. > * I cannot dial the DID and ring a phone. It only goes to VoiceMail. > > *Question:* > > * Why can I not dial the DID and have the extension ring? > * Where should I check? > > Thank you, > Mike > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130519/05b2ce03/attachment-0001.html From steveayre at gmail.com Mon May 20 10:20:58 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 20 May 2013 07:20:58 +0100 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <51995500.2030504@freeswitch.org> References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> <51995500.2030504@freeswitch.org> Message-ID: Sorry I meant to say with the daemon. (uuidd). That would be unique between multiple instances too. If the time one is used its already unique to the server. But as I said, I'm not sure if that's the one FS uses. On Monday, May 20, 2013, Yossi Neiman wrote: > If you're running multiple FS instances, you could also use both the FS > server UUID in addition to the session UUID together. That'd mean > storing more data but it would probably be on the order of winning a > $500 million USD Powerball game twice within the same month... > > -Yossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/7a58d05a/attachment.html From mike at hendrienet.com Mon May 20 07:39:41 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Sun, 19 May 2013 22:39:41 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: <5199827A.5080003@ppmt.org> References: <5199827A.5080003@ppmt.org> Message-ID: Thank you for your response. Per your request here is the result of the sofia status command: freeswitch at internal> sofia status Name Type Data State ================================================================================================= 10.1.1.5 alias internal ALIASED internal profile sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) external profile sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom NOREG GothamCity.xom alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) ================================================================================================= 3 profiles 2 aliases ========================================== /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml ========================================== /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml ========================================== /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml ========================================== /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml ========================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130519/efc38cbf/attachment.html From vermeulen.deon at gmail.com Mon May 20 15:00:35 2013 From: vermeulen.deon at gmail.com (Deon Vermeulen) Date: Mon, 20 May 2013 12:00:35 +0100 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform Message-ID: <519A0253.3060009@gmail.com> Hi, I've been looking at the Kazoo as a very viable option as a HostedPBX platform, but don't find any information with regards to Billing. I've also been tracking the thread about recommended GUI where 2600Hz are in the process for updated the code in git for Blue.box. This also looks like a possibility for a HostedPBX platform, but then again I have no idea about the Billing software to use for this. FusionPBX is a top contender if no the number 1 for a HostedPBX platform but again information for Billing is lacking. I would like to know what the feel out there is for Billing on a HostedPBX Platform, i.e. Cloud. If would be great if there is an opensource Billing system. I've purchased the jBilling Telecom guide, but no where can I find information about integrating with FreeSWITCH nor implementing it for a HostedPBX platform in the Cloud. Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/ab01e56d/attachment-0001.html From mike at jerris.com Mon May 20 16:18:01 2013 From: mike at jerris.com (Michael Jerris) Date: Mon, 20 May 2013 08:18:01 -0400 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> <51995500.2030504@freeswitch.org> Message-ID: <1B994D4B-F3E1-4492-85FB-26610F19BF49@jerris.com> The apr uuid stuff already has a portion of the uuid thats per-instance. I've seen a lot of details in this thread that are not accurate to my recollection of this code. Has anyone actually dug in and looked at it? Its been a while by my recollection is each instance gets a certain number of bits from random at startup, then its a combination of time + counter. The only collision possibility would be the random at startup portion which would be much less likely than getting struck by lightning. Mike On May 20, 2013, at 2:20 AM, Steven Ayre wrote: > Sorry I meant to say with the daemon. (uuidd). That would be unique between multiple instances too. If the time one is used its already unique to the server. But as I said, I'm not sure if that's the one FS uses. > > On Monday, May 20, 2013, Yossi Neiman wrote: > If you're running multiple FS instances, you could also use both the FS > server UUID in addition to the session UUID together. That'd mean > storing more data but it would probably be on the order of winning a > $500 million USD Powerball game twice within the same month... > > -Yossi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/9262f097/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon May 20 17:54:16 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 20 May 2013 14:54:16 +0100 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: <519A0253.3060009@gmail.com> References: <519A0253.3060009@gmail.com> Message-ID: Take a look at chargify.com - they are extremely good. Cal On Mon, May 20, 2013 at 12:00 PM, Deon Vermeulen wrote: > Hi, > > > I've been looking at the Kazoo as a very viable option as a HostedPBX > platform, but don't find any information with regards to Billing. > > I've also been tracking the thread about recommended GUI where 2600Hz are > in the process for updated the code in git for Blue.box. > This also looks like a possibility for a HostedPBX platform, but then > again I have no idea about the Billing software to use for this. > > FusionPBX is a top contender if no the number 1 for a HostedPBX platform > but again information for Billing is lacking. > > I would like to know what the feel out there is for Billing on a HostedPBX > Platform, i.e. Cloud. > > If would be great if there is an opensource Billing system. > > I've purchased the jBilling Telecom guide, but no where can I find > information about integrating with FreeSWITCH nor implementing it for a > HostedPBX platform in the Cloud. > > > Kind Regards > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/d5e3caf6/attachment.html From rnbrady at gmail.com Mon May 20 19:43:33 2013 From: rnbrady at gmail.com (Richard Brady) Date: Mon, 20 May 2013 16:43:33 +0100 Subject: [Freeswitch-users] sip_append_audio_sdp Message-ID: Hi folks I am having some trouble with the sip_append_audio_sdp variable. If I set it like: It seems to have absolutely no effect. I'm pretty sure I have seen it working in the past, so now I'm wondering what the prerequisites are for it to work (i.e. what am I doing wrong?!). I am using G729 in pass-through mode and have: inbound-late-negotiation [true] inbound-codec-negotiation [generous] The expected behaviour is for the value of this variable to be appended to the local SDP on the B-leg. The actual behaviour is that nothing is appended. Any thoughts? Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/9348a722/attachment.html From adahary at gmail.com Mon May 20 20:04:21 2013 From: adahary at gmail.com (adahary) Date: Mon, 20 May 2013 09:04:21 -0700 (PDT) Subject: [Freeswitch-users] directory sql tables Message-ID: <1369065861493-7590872.post@n2.nabble.com> I planning to switch from the directory xml files to sql tables using mod_xml_curl.I've already defined ODBC/Mysql and got all core sql tables installed.I've search for some sql tables for directory with all params/variables but couldn't find any.With Astwerisk RT there is a clear WiKi which layout all sql tables with user/system fields.Are there any ready to use directory sql tables? OR should I make my own tablesRegardsAssaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/directory-sql-tables-tp7590872.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/274242b2/attachment.html From bdfoster at davri.com Mon May 20 20:36:50 2013 From: bdfoster at davri.com (Brian Foster) Date: Mon, 20 May 2013 12:36:50 -0400 Subject: [Freeswitch-users] directory sql tables In-Reply-To: <1369065861493-7590872.post@n2.nabble.com> References: <1369065861493-7590872.post@n2.nabble.com> Message-ID: mod_xml_curl allows you to serve directory information from an outside web service. There will be no tables in freeswitch core db related to directory. You will need to develop your own service and db schema in order to serve directory using this module. There are several examples on the wiki and google. -BDF On May 20, 2013 12:09 PM, "adahary" wrote: > I planning to switch from the directory xml files to sql tables using > mod_xml_curl. I've already defined ODBC/Mysql and got all core sql tables > installed. I've search for some sql tables for directory with all > params/variables but couldn't find any. With Astwerisk RT there is a clear > WiKi which layout all sql tables with user/system fields. Are there any > ready to use directory sql tables? OR should I make my own tables Regards > Assaf > ------------------------------ > View this message in context: directory sql tables > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/30585f1e/attachment-0001.html From bdfoster at davri.com Mon May 20 20:43:15 2013 From: bdfoster at davri.com (Brian Foster) Date: Mon, 20 May 2013 12:43:15 -0400 Subject: [Freeswitch-users] Recommendation for Billing for Hosted PBX Platform In-Reply-To: References: <519A0253.3060009@gmail.com> Message-ID: Jbilling uses CDRs and events to do mediation. So using your guide, you should have an idea of what info is required from freeswitch. If you can give us some examples of what jbilling is asking for and we can give you ideas. Please start a new thread specifically for your jbilling issues. Thanks, - BDF On May 20, 2013 10:02 AM, "Cal Leeming [Simplicity Media Ltd]" < cal.leeming at simplicitymedialtd.co.uk> wrote: > Take a look at chargify.com - they are extremely good. > > Cal > > On Mon, May 20, 2013 at 12:00 PM, Deon Vermeulen > wrote: > >> Hi, >> >> >> I've been looking at the Kazoo as a very viable option as a HostedPBX >> platform, but don't find any information with regards to Billing. >> >> I've also been tracking the thread about recommended GUI where 2600Hz are >> in the process for updated the code in git for Blue.box. >> This also looks like a possibility for a HostedPBX platform, but then >> again I have no idea about the Billing software to use for this. >> >> FusionPBX is a top contender if no the number 1 for a HostedPBX platform >> but again information for Billing is lacking. >> >> I would like to know what the feel out there is for Billing on a >> HostedPBX Platform, i.e. Cloud. >> >> If would be great if there is an opensource Billing system. >> >> I've purchased the jBilling Telecom guide, but no where can I find >> information about integrating with FreeSWITCH nor implementing it for a >> HostedPBX platform in the Cloud. >> >> >> Kind Regards >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/2907d70d/attachment.html From adahary at gmail.com Mon May 20 21:16:21 2013 From: adahary at gmail.com (adahary) Date: Mon, 20 May 2013 10:16:21 -0700 (PDT) Subject: [Freeswitch-users] directory sql tables In-Reply-To: <1369065861493-7590872.post@n2.nabble.com> References: <1369065861493-7590872.post@n2.nabble.com> Message-ID: <1369070181716-7590875.post@n2.nabble.com> I'm aware of WiKi's that explain how to setup the curl/odbc links which I've already folloed up. I'll build then my own tables. thanks Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/directory-sql-tables-tp7590872p7590875.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mishehu at freeswitch.org Mon May 20 23:39:47 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Mon, 20 May 2013 14:39:47 -0500 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <1B994D4B-F3E1-4492-85FB-26610F19BF49@jerris.com> References: <5198DA65.20201@telecube.com.au> <006901ce54a2$7ec81500$7c583f00$@v10networks.ca> <51995500.2030504@freeswitch.org> <1B994D4B-F3E1-4492-85FB-26610F19BF49@jerris.com> Message-ID: <519A7C03.207@freeswitch.org> RE: the per session uuid: I've looked at the code recently, and I already don't recall the exact details. :-) However, if you have libuuid installed, it (apr and/or apr-util) defaults to using libuuid's functions and acts as nothing more than a shim. I think part of the computation is based upon /dev/urandom and part on the timestamp and possibly a counter, but I might be merging things in my head incorrectly as that might be the non-libuuid segment of the code... It's highly random, but as with any finite set, there is always a remote possibility of a collision. The FS core UUID is instantiated at start-up of FS, so every time FS is restarted, it gets a new FS core UUID. -Yossi On 05/20/2013 07:18 AM, Michael Jerris wrote: > The apr uuid stuff already has a portion of the uuid thats > per-instance. I've seen a lot of details in this thread that are not > accurate to my recollection of this code. Has anyone actually dug in > and looked at it? Its been a while by my recollection is each > instance gets a certain number of bits from random at startup, then > its a combination of time + counter. The only collision possibility > would be the random at startup portion which would be much less likely > than getting struck by lightning. > > Mike > > On May 20, 2013, at 2:20 AM, Steven Ayre > wrote: > >> Sorry I meant to say with the daemon. (uuidd). That would be unique >> between multiple instances too. If the time one is used its already >> unique to the server. But as I said, I'm not sure if that's the one >> FS uses. >> >> On Monday, May 20, 2013, Yossi Neiman wrote: >> >> If you're running multiple FS instances, you could also use both >> the FS >> server UUID in addition to the session UUID together. That'd mean >> storing more data but it would probably be on the order of winning a >> $500 million USD Powerball game twice within the same month... >> >> -Yossi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/62b04eec/attachment.html From chris.aloi at gmail.com Tue May 21 00:25:25 2013 From: chris.aloi at gmail.com (Christopher Aloi) Date: Mon, 20 May 2013 16:25:25 -0400 Subject: [Freeswitch-users] Notification of an event in the dialplan Message-ID: I'm currently using FreeSwitch between two systems as a "rate limiter' with the limit application. I bring calls in, run through a hash to identify a loop - if my hash variable is met, I transfer the call to a new extension where I log the event and return a 503 to the source. If not I return a 302 to the source to send the call on its way. I'd like a way to identify these events to our NOC via email or a SNMP trap but I am not sure the best way to do it. The alert would fall into the 503 extension. What is the best way to accomplish this? I'd like to keep the call flow as simple as possible. Any ideas? Thanks - - Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/1580799a/attachment-0001.html From bdfoster at davri.com Tue May 21 01:11:25 2013 From: bdfoster at davri.com (Brian Foster) Date: Mon, 20 May 2013 17:11:25 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> Message-ID: 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 mode may not work on a transcoded call. You shouldn't be transcoding if you can help it. Now, I'm not sure if that is an empty threat but you should enable late codec negotiation. Information can be found here: http://wiki.freeswitch.org/wiki/Codec_negotiation -BDF On May 17, 2013 3:48 AM, wrote: > i'm so stupid :) > tks > > http://pastebin.freeswitch.org/20933 > > called num is 022206... and when i try to use dtmf touch 5 it's not > works. > > tks > > Le 2013-05-17 09:25, Ken Rice a ?crit : > > it tells you the password in the popup... this is an anti spam thing > > > > KenSent from my iPad > > > > On May 17, 2013, at 0:46, Hermouet Erwan > > wrote: > > > >> On login i try my email...but don t work...i loose here > >> > >> Michael Collins a ?crit : > >> > >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet > >>> wrote: > >>> > >>>> I have the log but i never found how works pastebin ?? do you > >>>> have tutorial ? > >>> > >>> There isn't a tutorial. You log on, paste your stuff into the text > >>> box, select FreeSWITCH Log as the syntax highlighting and then > >>> click Send. Copy the URL from the browse address bar. it will be > >>> something like: > >>> http://pastebin.freeswitch.org/20927 [2] > >>> > >>> -MC > >> > >> Hermouet Erwan > >> Responsable technique > >> Bluetel > > > >> > > > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org [4] > >> http://www.freeswitchsolutions.com [5] > >> > >> > >> [6] > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org [7] > >> http://wiki.freeswitch.org [8] > >> http://www.cluecon.com [9] > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org [10] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [11] > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [12] > >> http://www.freeswitch.org [13] > > > > > > Links: > > ------ > > [1] mailto:ehermouet at bluetel.fr > > [2] http://pastebin.freeswitch.org/20927 > > [3] mailto:msc at freeswitch.org > > [4] mailto:consulting at freeswitch.org > > [5] http://www.freeswitchsolutions.com > > [6] > > [7] http://www.freeswitch.org > > [8] http://wiki.freeswitch.org > > [9] http://www.cluecon.com > > [10] mailto:FreeSWITCH-users at lists.freeswitch.org > > [11] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [13] http://www.freeswitch.org > > [14] mailto:ehermouet at bluetel.fr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/210973b9/attachment.html From netcentrica at gmail.com Tue May 21 01:19:42 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Mon, 20 May 2013 23:19:42 +0200 Subject: [Freeswitch-users] FreeSWITCH RTMFP support Message-ID: Hi Are there any plans to add support form Adobe RTMFP protocol? I know that there is mod_rtmp, but it supports only RTMP which is TCP-based. RTMFP is UDP based and designed specially for P2P communications and should give much smaller delays Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/2d9a3523/attachment.html From msc at freeswitch.org Tue May 21 01:46:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 20 May 2013 14:46:37 -0700 Subject: [Freeswitch-users] Notification of an event in the dialplan In-Reply-To: References: Message-ID: A simple approach might be to use the system app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system If you have a simple shell script that launches an email that would a quick and easy way to get going. -MC On Mon, May 20, 2013 at 1:25 PM, Christopher Aloi wrote: > > I'm currently using FreeSwitch between two systems as a "rate limiter' > with the limit application. > > I bring calls in, run through a hash to identify a loop - if my hash > variable is met, I transfer the call to a new extension where I log the > event and return a 503 to the source. If not I return a 302 to the source > to send the call on its way. > > I'd like a way to identify these events to our NOC via email or a SNMP > trap but I am not sure the best way to do it. > > The alert would fall into the 503 extension. > > What is the best way to accomplish this? > > I'd like to keep the call flow as simple as possible. > > Any ideas? > > Thanks - > > - Chris > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/40c1551d/attachment.html From philippe at ppmt.org Tue May 21 01:49:53 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Mon, 20 May 2013 17:49:53 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> Message-ID: <519A9A81.7020609@ppmt.org> I am no expert so I could be wrong but: First your outbound is no registered (NOREG) so that is most likely going to prevent outgoing call Also you refer to default_gateway in your dialplan....what value is it set to? Check vars.xmls I normally use directly the name of my gateway instead of using variable (might not be the best way though ) But you have the On 13-05-19 11:39 PM, Mike Hendrie wrote: > Thank you for your response. Per your request here is the result of > the sofia status command: > > freeswitch at internal> sofia status > Name Type Data > State > ================================================================================================= > 10.1.1.5 alias internal > ALIASED > internal profile sip:mod_sofia at 10.1.1.5:5060 > RUNNING (0) > external profile sip:mod_sofia at 10.1.1.5:5080 > RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom > REGED > external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom > NOREG > GothamCity.xom alias internal > ALIASED > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > ================================================================================================= > 3 profiles 2 aliases > > > ========================================== > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > > > > ========================================== > > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > ========================================== > /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ========================================== > /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > ========================================== > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130520/de6fd6f7/attachment-0001.html From chris.aloi at gmail.com Tue May 21 03:22:17 2013 From: chris.aloi at gmail.com (Christopher Aloi) Date: Mon, 20 May 2013 19:22:17 -0400 Subject: [Freeswitch-users] Notification of an event in the dialplan In-Reply-To: References: Message-ID: I was thinking about that - my concern was "and waits for the result". Could that potentially hang call processing? If I pass to a python -> sendmail script and wait for the result, if my script hangs, I would have these calls waiting for the result (right?) On Monday, May 20, 2013, Michael Collins wrote: > A simple approach might be to use the system app: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system > > If you have a simple shell script that launches an email that would a > quick and easy way to get going. > > -MC > > > On Mon, May 20, 2013 at 1:25 PM, Christopher Aloi > > wrote: > >> >> I'm currently using FreeSwitch between two systems as a "rate limiter' >> with the limit application. >> >> I bring calls in, run through a hash to identify a loop - if my hash >> variable is met, I transfer the call to a new extension where I log the >> event and return a 503 to the source. If not I return a 302 to the source >> to send the call on its way. >> >> I'd like a way to identify these events to our NOC via email or a SNMP >> trap but I am not sure the best way to do it. >> >> The alert would fall into the 503 extension. >> >> What is the best way to accomplish this? >> >> I'd like to keep the call flow as simple as possible. >> >> Any ideas? >> >> Thanks - >> >> - Chris >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org > 'consulting at freeswitch.org');> >> http://www.freeswitchsolutions.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 > 'FreeSWITCH-users at lists.freeswitch.org');> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/0c6fdbd3/attachment.html From marketing at cluecon.com Tue May 21 03:35:39 2013 From: marketing at cluecon.com (Michael Collins) Date: Mon, 20 May 2013 16:35:39 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello! The FreeSWITCH team is busy on several fronts but the most exciting news at the moment is that Packt Publishing has told us the new bookis officially off to production! If all goes well it should be available in the next few weeks. (It looks like they changed the cover art - once we know for sure what's on the cover we'll give it a nickname like we did with the "bridge" book.) Last week we had an impromptu discussionon various FreeSWITCH topics. We have rescheduled the VoIPMonitor.org conference call presentation for May 29th. Also, we have tentatively scheduled the HOMER presentation for June 5. That means we will have another open discussionfor this Wednesday's conference call. As a tip, if you would like to browse around and see what has been committed in FreeSWITCH lately you can use the Fisheye site. Here you can browse the source code and look at recent commits to see what has changed. If you prefer to look at the commits without the source code browser then try out the HTML front-end for our FreeSWITCH git repository. We also have repositories for contributions and sample configuration sets. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/377e41b1/attachment.html From leonardo.bidinoto at voicetechnology.com.br Tue May 21 05:33:48 2013 From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto) Date: Mon, 20 May 2013 22:33:48 -0300 Subject: [Freeswitch-users] mod_unimrcp dont send invite while using a Mrcp v2 profile Message-ID: Hi All, Im new at using the mod_unimrcp. I'm having some troubles when configuring to using several profiles at same time. I have configured 3 profiles(2 with Nuance and 1 Verbio, all using MRCPv2). But when i try to use a TTS or ASR from any of those profile, FS channel get stuck and i get a warning message telling that the time for the MRCP server to respond have expired. I have used wireshark to capture the packages, and when i call speak or detect_speech function of FS, i dont see any SIP INVITE going from FS to the MRCP Server. Did i miss something at configuring profiles? Did anyone get something similar to this situation? I'm using the examples from FS mod_unimrcp page. Any tip will help a lot. Thanks -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/9b9c86b3/attachment.html From gabe at gundy.org Tue May 21 06:58:35 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 20 May 2013 20:58:35 -0600 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <5198DA65.20201@telecube.com.au> References: <5198DA65.20201@telecube.com.au> Message-ID: On Sun, May 19, 2013 at 7:57 AM, Telecube - John wrote: > Can I count on uuid to be totally unique in my billing records? Just for fun... Number of stars believed to exist in known universe: 70,000,000,000,000,000,000,000 Number of *drops* of water estimated to be the world's oceans (U.S. Navy): 222,000,000,000,000,000,000,000,000 Number of *unique* UUIDs: 340,282,366,920,938,463,463,374,607,431,768,211,456 So, your chances of a collision with *hundreds* of *billions* of truly random UUIDs is inconceivably low. The only thing you really have to worry about are humans and the code they write ;) Best, Gabe From john at telecube.com.au Tue May 21 07:16:13 2013 From: john at telecube.com.au (Telecube - John) Date: Tue, 21 May 2013 13:16:13 +1000 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: References: <5198DA65.20201@telecube.com.au> Message-ID: <519AE6FD.8020205@telecube.com.au> +rep I like this answer :-) - John On 21/05/13 12:58 PM, Gabriel Gunderson wrote: > On Sun, May 19, 2013 at 7:57 AM, Telecube - John wrote: >> Can I count on uuid to be totally unique in my billing records? > Just for fun... > > Number of stars believed to exist in known universe: > 70,000,000,000,000,000,000,000 > > Number of *drops* of water estimated to be the world's oceans (U.S. Navy): > 222,000,000,000,000,000,000,000,000 > > Number of *unique* UUIDs: > 340,282,366,920,938,463,463,374,607,431,768,211,456 > > So, your chances of a collision with *hundreds* of *billions* of truly > random UUIDs is inconceivably low. > > The only thing you really have to worry about are humans and the code > they write ;) > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From mishehu at freeswitch.org Tue May 21 08:19:52 2013 From: mishehu at freeswitch.org (Yossi Neiman) Date: Mon, 20 May 2013 23:19:52 -0500 Subject: [Freeswitch-users] FreeSWITCH RTMFP support In-Reply-To: References: Message-ID: <519AF5E8.4040909@freeswitch.org> The problem with RTMFP is that is it a closed protocol, and has no official library implementation available. Of the available reverse-engineered libraries, I do not know of one that is license-compatible with FreeSWITCH. If you happen to know of one, please let us know. There are companies that purport to provide RTMFP-to-SIP gateway software, but I cannot vouch for their viability. -Yossi From dujinfang at gmail.com Tue May 21 08:43:27 2013 From: dujinfang at gmail.com (Seven Du) Date: Tue, 21 May 2013 12:43:27 +0800 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: <519AE6FD.8020205@telecube.com.au> References: <5198DA65.20201@telecube.com.au> <519AE6FD.8020205@telecube.com.au> Message-ID: <59D70102867D4554AB2E24F7A951CF67@gmail.com> I had looked this on wikipedia a few days ago, it said - In other words, only after generating 1 billion UUIDs every second for the next 100 years, the probability of creating just one duplicate would be about 50%. http://zh.wikipedia.org/wiki/UUID -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, May 21, 2013 at 11:16 AM, Telecube - John wrote: > +rep > > I like this answer > > :-) > > - John > On 21/05/13 12:58 PM, Gabriel Gunderson wrote: > > On Sun, May 19, 2013 at 7:57 AM, Telecube - John wrote: > > > Can I count on uuid to be totally unique in my billing records? > > > > Just for fun... > > > > Number of stars believed to exist in known universe: > > 70,000,000,000,000,000,000,000 > > > > Number of *drops* of water estimated to be the world's oceans (U.S. Navy): > > 222,000,000,000,000,000,000,000,000 > > > > Number of *unique* UUIDs: > > 340,282,366,920,938,463,463,374,607,431,768,211,456 > > > > So, your chances of a collision with *hundreds* of *billions* of truly > > random UUIDs is inconceivably low. > > > > The only thing you really have to worry about are humans and the code > > they write ;) > > > > > > Best, > > Gabe > > > > _________________________________________________________________________ > > 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://wiki.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://wiki.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/20130521/3146f925/attachment-0001.html From avi at avimarcus.net Tue May 21 09:40:40 2013 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 21 May 2013 08:40:40 +0300 Subject: [Freeswitch-users] Notification of an event in the dialplan In-Reply-To: References: Message-ID: Hmm, this looks promising: http://wiki.freeswitch.org/wiki/Mod_commands#bg_system Execute a system command in the background. Usage: bg_system Please update the wiki if you get more information about how that works... -Avi On Tue, May 21, 2013 at 2:22 AM, Christopher Aloi wrote: > I was thinking about that - my concern was "and waits for the result". > Could that potentially hang call processing? If I pass to a python -> > sendmail script and wait for the result, if my script hangs, I would have > these calls waiting for the result (right?) > > On Monday, May 20, 2013, Michael Collins wrote: > >> A simple approach might be to use the system app: >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system >> >> If you have a simple shell script that launches an email that would a >> quick and easy way to get going. >> >> -MC >> >> >> On Mon, May 20, 2013 at 1:25 PM, Christopher Aloi wrote: >> >>> >>> I'm currently using FreeSwitch between two systems as a "rate limiter' >>> with the limit application. >>> >>> I bring calls in, run through a hash to identify a loop - if my hash >>> variable is met, I transfer the call to a new extension where I log the >>> event and return a 503 to the source. If not I return a 302 to the source >>> to send the call on its way. >>> >>> I'd like a way to identify these events to our NOC via email or a SNMP >>> trap but I am not sure the best way to do it. >>> >>> The alert would fall into the 503 extension. >>> >>> What is the best way to accomplish this? >>> >>> I'd like to keep the call flow as simple as possible. >>> >>> Any ideas? >>> >>> Thanks - >>> >>> - Chris >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/8f9aae8b/attachment.html From ostolyar at netflix.com Tue May 21 06:10:09 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Mon, 20 May 2013 19:10:09 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy Message-ID: Hi, I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below which simply sets bypass_media to true and then bridges to another FreeSWITCH server. However, when during the call I shut down the proxy FS, the call is immediately dropped. Why is that? Is there a way to keep it going? I understand that in this case I won't be able to properly send the BYE signal when one of the parties hangs up and that's OK. I tried using redirect and deflect instead of the bridge but those don't seem to work at all - probably because my UAs don't know how to handle redirects. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130520/d4c2f756/attachment-0001.html From mike at hendrienet.com Tue May 21 08:50:52 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Mon, 20 May 2013 23:50:52 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: <519A9A81.7020609@ppmt.org> References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: I will take a closer look at the gateways. However, your email looks like it was sent before you completed it. What about the inbound DID? Any ideas? On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: > I am no expert so I could be wrong but: > > First your outbound is no registered (NOREG) so that is most likely going > to prevent outgoing call > > Also you refer to default_gateway in your dialplan....what value is it set > to? Check vars.xmls > > I normally use directly the name of my gateway instead of using variable > (might not be the best way though ) > > > > But you have the > On 13-05-19 11:39 PM, Mike Hendrie wrote: > > Thank you for your response. Per your request here is the result of the sofia > status command: > > freeswitch at internal> sofia status > Name Type > Data State > > ================================================================================================= > 10.1.1.5 alias > internal ALIASED > internal profile > sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) > external profile > sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom > REGED > external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom > NOREG > GothamCity.xom alias > internal ALIASED > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > > ================================================================================================= > 3 profiles 2 aliases > > > ========================================== > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > > > > ========================================== > > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > ========================================== > /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ========================================== > /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > ========================================== > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130520/c8249b4b/attachment-0001.html From oej at edvina.net Tue May 21 10:50:26 2013 From: oej at edvina.net (Olle E. Johansson) Date: Tue, 21 May 2013 08:50:26 +0200 Subject: [Freeswitch-users] UUID - how unique is it? In-Reply-To: References: <5198DA65.20201@telecube.com.au> Message-ID: 21 maj 2013 kl. 04:58 skrev Gabriel Gunderson : > The only thing you really have to worry about are humans and the code > they write ;) > Now I got really scared. /O ;-) From xiaofengcanyuexp at 163.com Tue May 21 13:41:43 2013 From: xiaofengcanyuexp at 163.com (windy) Date: Tue, 21 May 2013 17:41:43 +0800 Subject: [Freeswitch-users] custom sip header not working in dialplan References: <201305211732533444389@163.com> Message-ID: <201305211741427839968@163.com> Hi, Freeswitch Team There is problem for to modify the outbound sip header. As the freeswitch wiki says, it can modify/add custom sip header via below method: Now, I add the red lines in dialplan/default.xml. But only the custom header prefixed with "sip_h_" can be shown in the SIP INVITE message. Other custom headers prefixed with "sip_bye_h" and "sip_ph"... not found in the related messages(BYE, 183). Could you kindly give comments about this case? Thanks Windy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/75dcbb44/attachment.html From juanito1982 at gmail.com Tue May 21 14:51:27 2013 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 21 May 2013 12:51:27 +0200 Subject: [Freeswitch-users] GXP1405 G729 audio problem Message-ID: Hello, Anyone had problems using G279 codec with Grandstream GXP1405. GXP1405 is connected to FS which is connected to an Asterisk box. FS has no codec installed so it is in passthrough mode. The audio received in the phone is bad. Sometimes volume goes down and it has low quality. Working connected directly to Asterisk using G729 box goes well. Working through FS using G711 goes well. If I analyze both flows, from Asterisk and from FS, inside FS box using G729, all audio is OK. So I don't know why phone plays it bad... Any idea? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/7473dc4b/attachment.html From ehermouet at bluetel.fr Tue May 21 15:43:35 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Tue, 21 May 2013 13:43:35 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> Message-ID: <8cdce6d356b9217fc814e3fc12f1933c@bluetel.fr> Tks for your reply, now i don't have error on log but, dtmf is not send again. please i need quick help tks advance Le 2013-05-20 23:11, Brian Foster a ?crit?: > 2013-05-17?09:39:54.214928?[WARNING]?mod_sofia.c:1363?Pass?2833?mode > may not work on a transcoded call. > > You shouldnt be transcoding if you can help it. Now, Im not sure if > that is an empty threat but you should enable late codec negotiation. > Information can be found here: > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] > > -BDF > On May 17, 2013 3:48 AM, wrote: > >> im so stupid :) >> tks >> >> http://pastebin.freeswitch.org/20933 [1] >> >> called num is 022206... and when i try to use dtmf touch 5 its not >> works. >> >> tks >> >> Le 2013-05-17 09:25, Ken Rice a ?crit?: >> > it tells you the password in the popup... this is an anti spam >> thing >> > >> > KenSent from my iPad >> > >> > On May 17, 2013, at 0:46, Hermouet Erwan > [2] [14]> >> > wrote: >> > >> >> On login i try my email...but don t work...i loose here >> >> >> >> Michael Collins a ?crit : >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >>> wrote: >> >>> >> >>>> I have the log but i never found how works pastebin ?? do you >> >>>> have tutorial ? >> >>> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> text >> >>> box, select FreeSWITCH Log as the syntax highlighting and then >> >>> click Send. Copy the URL from the browse address bar. it will >> be >> >>> something like: >> >>> http://pastebin.freeswitch.org/20927 [5] [2] >> >>> >> >>> -MC >> >> >> >> Hermouet Erwan >> >> Responsable technique >> >> Bluetel >> > >> >> >> > >> > >> > > _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [6] [4] >> >> http://www.freeswitchsolutions.com [7] [5] >> >> >> >> >> >> [8] [6] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [9] [7] >> >> http://wiki.freeswitch.org [10] [8] >> >> http://www.cluecon.com [11] [9] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [13] [11] >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [14] >> >> [12] >> >> http://www.freeswitch.org [15] [13] >> > >> > >> > Links: >> > ------ >> > [1] mailto:ehermouet at bluetel.fr [16] >> > [2] http://pastebin.freeswitch.org/20927 [17] >> > [3] mailto:msc at freeswitch.org [18] >> > [4] mailto:consulting at freeswitch.org [19] >> > [5] http://www.freeswitchsolutions.com [20] >> > [6] [21] >> > [7] http://www.freeswitch.org [22] >> > [8] http://wiki.freeswitch.org [23] >> > [9] http://www.cluecon.com [24] >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] >> > [11] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] >> > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [27] >> > [13] http://www.freeswitch.org [28] >> > [14] mailto:ehermouet at bluetel.fr [29] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [30] >> http://www.freeswitchsolutions.com [31] >> >> >> [32] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [33] >> http://wiki.freeswitch.org [34] >> http://www.cluecon.com [35] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [36] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [38] >> http://www.freeswitch.org [39] > > > Links: > ------ > [1] http://pastebin.freeswitch.org/20933 > [2] mailto:ehermouet at bluetel.fr > [3] mailto:msc at freeswitch.org > [4] mailto:ehermouet at bluetel.fr > [5] http://pastebin.freeswitch.org/20927 > [6] mailto:consulting at freeswitch.org > [7] http://www.freeswitchsolutions.com > [8] > [9] http://www.freeswitch.org > [10] http://wiki.freeswitch.org > [11] http://www.cluecon.com > [12] mailto:FreeSWITCH-users at lists.freeswitch.org > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users > [15] http://www.freeswitch.org > [16] mailto:ehermouet at bluetel.fr > [17] http://pastebin.freeswitch.org/20927 > [18] mailto:msc at freeswitch.org > [19] mailto:consulting at freeswitch.org > [20] http://www.freeswitchsolutions.com > [21] > [22] http://www.freeswitch.org > [23] http://wiki.freeswitch.org > [24] http://www.cluecon.com > [25] mailto:FreeSWITCH-users at lists.freeswitch.org > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users > [28] http://www.freeswitch.org > [29] mailto:ehermouet at bluetel.fr > [30] mailto:consulting at freeswitch.org > [31] http://www.freeswitchsolutions.com > [32] > [33] http://www.freeswitch.org > [34] http://wiki.freeswitch.org > [35] http://www.cluecon.com > [36] mailto:FreeSWITCH-users at lists.freeswitch.org > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users > [39] http://www.freeswitch.org > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation > [41] mailto:ehermouet at bluetel.fr From ehermouet at bluetel.fr Tue May 21 15:53:46 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Tue, 21 May 2013 13:53:46 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> Message-ID: <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> I must use ? if yes where and how ? tks advance for your help Le 2013-05-20 23:11, Brian Foster a ?crit?: > 2013-05-17?09:39:54.214928?[WARNING]?mod_sofia.c:1363?Pass?2833?mode > may not work on a transcoded call. > > You shouldnt be transcoding if you can help it. Now, Im not sure if > that is an empty threat but you should enable late codec negotiation. > Information can be found here: > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] > > -BDF > On May 17, 2013 3:48 AM, wrote: > >> im so stupid :) >> tks >> >> http://pastebin.freeswitch.org/20933 [1] >> >> called num is 022206... and when i try to use dtmf touch 5 its not >> works. >> >> tks >> >> Le 2013-05-17 09:25, Ken Rice a ?crit?: >> > it tells you the password in the popup... this is an anti spam >> thing >> > >> > KenSent from my iPad >> > >> > On May 17, 2013, at 0:46, Hermouet Erwan > [2] [14]> >> > wrote: >> > >> >> On login i try my email...but don t work...i loose here >> >> >> >> Michael Collins a ?crit : >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >>> wrote: >> >>> >> >>>> I have the log but i never found how works pastebin ?? do you >> >>>> have tutorial ? >> >>> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> text >> >>> box, select FreeSWITCH Log as the syntax highlighting and then >> >>> click Send. Copy the URL from the browse address bar. it will >> be >> >>> something like: >> >>> http://pastebin.freeswitch.org/20927 [5] [2] >> >>> >> >>> -MC >> >> >> >> Hermouet Erwan >> >> Responsable technique >> >> Bluetel >> > >> >> >> > >> > >> > > _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [6] [4] >> >> http://www.freeswitchsolutions.com [7] [5] >> >> >> >> >> >> [8] [6] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [9] [7] >> >> http://wiki.freeswitch.org [10] [8] >> >> http://www.cluecon.com [11] [9] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [13] [11] >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [14] >> >> [12] >> >> http://www.freeswitch.org [15] [13] >> > >> > >> > Links: >> > ------ >> > [1] mailto:ehermouet at bluetel.fr [16] >> > [2] http://pastebin.freeswitch.org/20927 [17] >> > [3] mailto:msc at freeswitch.org [18] >> > [4] mailto:consulting at freeswitch.org [19] >> > [5] http://www.freeswitchsolutions.com [20] >> > [6] [21] >> > [7] http://www.freeswitch.org [22] >> > [8] http://wiki.freeswitch.org [23] >> > [9] http://www.cluecon.com [24] >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] >> > [11] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] >> > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [27] >> > [13] http://www.freeswitch.org [28] >> > [14] mailto:ehermouet at bluetel.fr [29] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [30] >> http://www.freeswitchsolutions.com [31] >> >> >> [32] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [33] >> http://wiki.freeswitch.org [34] >> http://www.cluecon.com [35] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [36] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [38] >> http://www.freeswitch.org [39] > > > Links: > ------ > [1] http://pastebin.freeswitch.org/20933 > [2] mailto:ehermouet at bluetel.fr > [3] mailto:msc at freeswitch.org > [4] mailto:ehermouet at bluetel.fr > [5] http://pastebin.freeswitch.org/20927 > [6] mailto:consulting at freeswitch.org > [7] http://www.freeswitchsolutions.com > [8] > [9] http://www.freeswitch.org > [10] http://wiki.freeswitch.org > [11] http://www.cluecon.com > [12] mailto:FreeSWITCH-users at lists.freeswitch.org > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users > [15] http://www.freeswitch.org > [16] mailto:ehermouet at bluetel.fr > [17] http://pastebin.freeswitch.org/20927 > [18] mailto:msc at freeswitch.org > [19] mailto:consulting at freeswitch.org > [20] http://www.freeswitchsolutions.com > [21] > [22] http://www.freeswitch.org > [23] http://wiki.freeswitch.org > [24] http://www.cluecon.com > [25] mailto:FreeSWITCH-users at lists.freeswitch.org > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users > [28] http://www.freeswitch.org > [29] mailto:ehermouet at bluetel.fr > [30] mailto:consulting at freeswitch.org > [31] http://www.freeswitchsolutions.com > [32] > [33] http://www.freeswitch.org > [34] http://wiki.freeswitch.org > [35] http://www.cluecon.com > [36] mailto:FreeSWITCH-users at lists.freeswitch.org > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users > [39] http://www.freeswitch.org > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation > [41] mailto:ehermouet at bluetel.fr From bdfoster at davri.com Tue May 21 16:02:05 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 08:02:05 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <8cdce6d356b9217fc814e3fc12f1933c@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <8cdce6d356b9217fc814e3fc12f1933c@bluetel.fr> Message-ID: Please pastebin another console log with sip trace on globally: sofia global siptrace on -BDF On May 21, 2013 7:49 AM, wrote: > Tks for your reply, > > now i don't have error on log but, dtmf is not send again. > > please i need quick help > > tks advance > > > Le 2013-05-20 23:11, Brian Foster a ?crit : > > 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 mode > > may not work on a transcoded call. > > > > You shouldnt be transcoding if you can help it. Now, Im not sure if > > that is an empty threat but you should enable late codec negotiation. > > Information can be found here: > > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] > > > > -BDF > > On May 17, 2013 3:48 AM, wrote: > > > >> im so stupid :) > >> tks > >> > >> http://pastebin.freeswitch.org/20933 [1] > >> > >> called num is 022206... and when i try to use dtmf touch 5 its not > >> works. > >> > >> tks > >> > >> Le 2013-05-17 09:25, Ken Rice a ?crit : > >> > it tells you the password in the popup... this is an anti spam > >> thing > >> > > >> > KenSent from my iPad > >> > > >> > On May 17, 2013, at 0:46, Hermouet Erwan >> [2] [14]> > >> > wrote: > >> > > >> >> On login i try my email...but don t work...i loose here > >> >> > >> >> Michael Collins a ?crit : > >> >> > >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet > >> >>> wrote: > >> >>> > >> >>>> I have the log but i never found how works pastebin ?? do you > >> >>>> have tutorial ? > >> >>> > >> >>> There isnt a tutorial. You log on, paste your stuff into the > >> text > >> >>> box, select FreeSWITCH Log as the syntax highlighting and then > >> >>> click Send. Copy the URL from the browse address bar. it will > >> be > >> >>> something like: > >> >>> http://pastebin.freeswitch.org/20927 [5] [2] > >> >>> > >> >>> -MC > >> >> > >> >> Hermouet Erwan > >> >> Responsable technique > >> >> Bluetel > >> > > >> >> > >> > > >> > > >> > > > > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org [6] [4] > >> >> http://www.freeswitchsolutions.com [7] [5] > >> >> > >> >> > >> >> [8] [6] > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org [9] [7] > >> >> http://wiki.freeswitch.org [10] [8] > >> >> http://www.cluecon.com [11] [9] > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> [13] [11] > >> >> > >> > > >> > > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [14] > >> >> [12] > >> >> http://www.freeswitch.org [15] [13] > >> > > >> > > >> > Links: > >> > ------ > >> > [1] mailto:ehermouet at bluetel.fr [16] > >> > [2] http://pastebin.freeswitch.org/20927 [17] > >> > [3] mailto:msc at freeswitch.org [18] > >> > [4] mailto:consulting at freeswitch.org [19] > >> > [5] http://www.freeswitchsolutions.com [20] > >> > [6] [21] > >> > [7] http://www.freeswitch.org [22] > >> > [8] http://wiki.freeswitch.org [23] > >> > [9] http://www.cluecon.com [24] > >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] > >> > [11] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] > >> > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [27] > >> > [13] http://www.freeswitch.org [28] > >> > [14] mailto:ehermouet at bluetel.fr [29] > >> > >> > > > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org [30] > >> http://www.freeswitchsolutions.com [31] > >> > >> > >> [32] > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org [33] > >> http://wiki.freeswitch.org [34] > >> http://www.cluecon.com [35] > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org [36] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [38] > >> http://www.freeswitch.org [39] > > > > > > Links: > > ------ > > [1] http://pastebin.freeswitch.org/20933 > > [2] mailto:ehermouet at bluetel.fr > > [3] mailto:msc at freeswitch.org > > [4] mailto:ehermouet at bluetel.fr > > [5] http://pastebin.freeswitch.org/20927 > > [6] mailto:consulting at freeswitch.org > > [7] http://www.freeswitchsolutions.com > > [8] > > [9] http://www.freeswitch.org > > [10] http://wiki.freeswitch.org > > [11] http://www.cluecon.com > > [12] mailto:FreeSWITCH-users at lists.freeswitch.org > > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [15] http://www.freeswitch.org > > [16] mailto:ehermouet at bluetel.fr > > [17] http://pastebin.freeswitch.org/20927 > > [18] mailto:msc at freeswitch.org > > [19] mailto:consulting at freeswitch.org > > [20] http://www.freeswitchsolutions.com > > [21] > > [22] http://www.freeswitch.org > > [23] http://wiki.freeswitch.org > > [24] http://www.cluecon.com > > [25] mailto:FreeSWITCH-users at lists.freeswitch.org > > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [28] http://www.freeswitch.org > > [29] mailto:ehermouet at bluetel.fr > > [30] mailto:consulting at freeswitch.org > > [31] http://www.freeswitchsolutions.com > > [32] > > [33] http://www.freeswitch.org > > [34] http://wiki.freeswitch.org > > [35] http://www.cluecon.com > > [36] mailto:FreeSWITCH-users at lists.freeswitch.org > > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [39] http://www.freeswitch.org > > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation > > [41] mailto:ehermouet at bluetel.fr > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/e31cc01f/attachment-0001.html From bdfoster at davri.com Tue May 21 16:03:26 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 08:03:26 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> Message-ID: No, you will not need to use that. On May 21, 2013 8:00 AM, wrote: > I must use ? if yes where > and how ? > > tks advance for your help > > > Le 2013-05-20 23:11, Brian Foster a ?crit : > > 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 mode > > may not work on a transcoded call. > > > > You shouldnt be transcoding if you can help it. Now, Im not sure if > > that is an empty threat but you should enable late codec negotiation. > > Information can be found here: > > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] > > > > -BDF > > On May 17, 2013 3:48 AM, wrote: > > > >> im so stupid :) > >> tks > >> > >> http://pastebin.freeswitch.org/20933 [1] > >> > >> called num is 022206... and when i try to use dtmf touch 5 its not > >> works. > >> > >> tks > >> > >> Le 2013-05-17 09:25, Ken Rice a ?crit : > >> > it tells you the password in the popup... this is an anti spam > >> thing > >> > > >> > KenSent from my iPad > >> > > >> > On May 17, 2013, at 0:46, Hermouet Erwan >> [2] [14]> > >> > wrote: > >> > > >> >> On login i try my email...but don t work...i loose here > >> >> > >> >> Michael Collins a ?crit : > >> >> > >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet > >> >>> wrote: > >> >>> > >> >>>> I have the log but i never found how works pastebin ?? do you > >> >>>> have tutorial ? > >> >>> > >> >>> There isnt a tutorial. You log on, paste your stuff into the > >> text > >> >>> box, select FreeSWITCH Log as the syntax highlighting and then > >> >>> click Send. Copy the URL from the browse address bar. it will > >> be > >> >>> something like: > >> >>> http://pastebin.freeswitch.org/20927 [5] [2] > >> >>> > >> >>> -MC > >> >> > >> >> Hermouet Erwan > >> >> Responsable technique > >> >> Bluetel > >> > > >> >> > >> > > >> > > >> > > > > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org [6] [4] > >> >> http://www.freeswitchsolutions.com [7] [5] > >> >> > >> >> > >> >> [8] [6] > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org [9] [7] > >> >> http://wiki.freeswitch.org [10] [8] > >> >> http://www.cluecon.com [11] [9] > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> [13] [11] > >> >> > >> > > >> > > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [14] > >> >> [12] > >> >> http://www.freeswitch.org [15] [13] > >> > > >> > > >> > Links: > >> > ------ > >> > [1] mailto:ehermouet at bluetel.fr [16] > >> > [2] http://pastebin.freeswitch.org/20927 [17] > >> > [3] mailto:msc at freeswitch.org [18] > >> > [4] mailto:consulting at freeswitch.org [19] > >> > [5] http://www.freeswitchsolutions.com [20] > >> > [6] [21] > >> > [7] http://www.freeswitch.org [22] > >> > [8] http://wiki.freeswitch.org [23] > >> > [9] http://www.cluecon.com [24] > >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] > >> > [11] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] > >> > [12] http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [27] > >> > [13] http://www.freeswitch.org [28] > >> > [14] mailto:ehermouet at bluetel.fr [29] > >> > >> > > > > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org [30] > >> http://www.freeswitchsolutions.com [31] > >> > >> > >> [32] > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org [33] > >> http://wiki.freeswitch.org [34] > >> http://www.cluecon.com [35] > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org [36] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] > >> > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> [38] > >> http://www.freeswitch.org [39] > > > > > > Links: > > ------ > > [1] http://pastebin.freeswitch.org/20933 > > [2] mailto:ehermouet at bluetel.fr > > [3] mailto:msc at freeswitch.org > > [4] mailto:ehermouet at bluetel.fr > > [5] http://pastebin.freeswitch.org/20927 > > [6] mailto:consulting at freeswitch.org > > [7] http://www.freeswitchsolutions.com > > [8] > > [9] http://www.freeswitch.org > > [10] http://wiki.freeswitch.org > > [11] http://www.cluecon.com > > [12] mailto:FreeSWITCH-users at lists.freeswitch.org > > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [15] http://www.freeswitch.org > > [16] mailto:ehermouet at bluetel.fr > > [17] http://pastebin.freeswitch.org/20927 > > [18] mailto:msc at freeswitch.org > > [19] mailto:consulting at freeswitch.org > > [20] http://www.freeswitchsolutions.com > > [21] > > [22] http://www.freeswitch.org > > [23] http://wiki.freeswitch.org > > [24] http://www.cluecon.com > > [25] mailto:FreeSWITCH-users at lists.freeswitch.org > > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [28] http://www.freeswitch.org > > [29] mailto:ehermouet at bluetel.fr > > [30] mailto:consulting at freeswitch.org > > [31] http://www.freeswitchsolutions.com > > [32] > > [33] http://www.freeswitch.org > > [34] http://wiki.freeswitch.org > > [35] http://www.cluecon.com > > [36] mailto:FreeSWITCH-users at lists.freeswitch.org > > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users > > [39] http://www.freeswitch.org > > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation > > [41] mailto:ehermouet at bluetel.fr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/946be38d/attachment.html From bdfoster at davri.com Tue May 21 16:07:18 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 08:07:18 -0400 Subject: [Freeswitch-users] Caller ID and Caller Name Issue In-Reply-To: References: Message-ID: Please pastebin your dialplans. Is this an outgoing call or inbound? -BDF On May 18, 2013 9:45 AM, "Jagadish Thoutam" wrote: > Hi Michael, > > i am not sure its strange because i tried everything what you > said and even i am working with freeswitch from last 2 years i didn't have > problem like this But still the same thing. > > callerid = FreeSWITCH > > caleerid name = +1XXXXXXXXXX > > > Thanks > Jagan > > > > On 17 May 2013 19:20, Michael Collins wrote: > >> >> >> >> On Fri, May 17, 2013 at 3:24 PM, Muhammad Shahzad wrote: >> >>> How are you disabling the global caller id? This won't work, (its a >>> common mistake done by many users when dealing with X-PRE-PROCESS) >>> >>> >>> >>> You have to do something like this, >>> >>> >>> >> FYI, we recommend not using XML comments when trying to disable a >> preprocess directive. Just do something like this: >> >> >> >> That will do the trick and will make the code easier to read. >> -MC >> >> >>> >>> Notice the space around ' - ' in X-PRE-PROCESS. >>> >>> Also you should set some appropriate caller id for outbound call in your >>> dial plan, as already i told you in my previous email. >>> >>> Thank you. >>> >>> >>> >>> >>> On Fri, May 17, 2013 at 10:26 PM, Jagadish Thoutam < >>> jaganthoutam at gmail.com> wrote: >>> >>>> Hi Muhammad, >>>> >>>> i disable the global outbound caller id, >>>> >>>> My setup is like this >>>> >>>> DID -----------> FreeSWITCH -------> AsteriskServer >>>> >>>> >>>> >>>> >>>> >>> expression="^(?:\+1)(88XXXXXXXX)$"> >>>> >>>> >>>> >>>> >>>> >>>> ASTERISK is my gateway in My Asterisk is getting FreeSWITCH as >>>> CallerID not evean as CallerNAme. >>>> >>>> >>>> >>>> >>>> >>>> Thanks >>>> Jagan >>>> >>>> >>>> >>>> >>>> >>>> On 17 May 2013 13:01, Muhammad Shahzad wrote: >>>> >>>>> Well, if you want to have variable caller id number and name then use >>>>> "effective_caller_id_name" and "effective_caller_id_number" channel >>>>> variables in your dial plan. Beispiel, >>>>> >>>>> >>>>> >>>> data="effective_caller_id_number=49185551212"/> >>>>> >>>>> OR if you want to use a global outbound caller id name and number then >>>>> you can modify following in vars.xml, >>>>> >>>>> >>>>> >>>>> >>>>> Thank you. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, May 17, 2013 at 5:45 PM, Jagadish Thoutam < >>>>> jaganthoutam at gmail.com> wrote: >>>>> >>>>>> Any help on this issue. >>>>>> >>>>>> sent from samsung S3 >>>>>> On 17 May 2013 11:32, "Jagadish Thoutam" >>>>>> wrote: >>>>>> >>>>>>> No, it didn't help me Still i am getting FreeSWITH as a caller ID >>>>>>> (not Caller ID Name) caller id Name i am getting Number. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Thanks >>>>>>> Jagan >>>>>>> >>>>>>> >>>>>>> On 16 May 2013 20:53, wrote: >>>>>>> >>>>>>>> Try on profile >>>>>>>> setting. >>>>>>>> >>>>>>>> On 16.05.2013 16:24, Jagadish Thoutam wrote: >>>>>>>> > HI, >>>>>>>> > >>>>>>>> > >>>>>>>> > i have a freeswitch as a VOIP Gateway that will handle the >>>>>>>> Inbound >>>>>>>> > and Outbound calls with media proxy, here issue comes My >>>>>>>> Freeswith is >>>>>>>> > Sending FreeSwitch as a callid Number.. and >>>>>>>> > callid name as a Number i wondered why it is sending like >>>>>>>> that >>>>>>>> > i even comment the effective acller id name and caller id but >>>>>>>> still >>>>>>>> > same issue, can any one help on this. >>>>>>>> > >>>>>>>> > Version : >>>>>>>> > >>>>>>>> > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git >>>>>>>> > e1c325d >>>>>>>> > 2013-04-23 19:49:07Z) >>>>>>>> > >>>>>>>> > Thanks >>>>>>>> > Jagan >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.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 >>>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Mit freundlichen Gr??en >>>>> Muhammad Shahzad >>>>> ----------------------------------- >>>>> CISCO Rich Media Communication Specialist (CRMCS) >>>>> CISCO Certified Network Associate (CCNA) >>>>> Cell: +49 176 99 83 10 85 >>>>> MSN: shari_786pk at hotmail.com >>>>> Email: shaheryarkh at googlemail.com >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Mit freundlichen Gr??en >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/d542c774/attachment-0001.html From ehermouet at bluetel.fr Tue May 21 16:36:27 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Tue, 21 May 2013 14:36:27 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <8cdce6d356b9217fc814e3fc12f1933c@bluetel.fr> Message-ID: <779e64193fa5f5d2b763b59be442ec5c@bluetel.fr> http://pastebin.freeswitch.org/20947 Tks Le 2013-05-21 14:02, Brian Foster a ?crit?: > Please pastebin another console log with sip trace on globally: > > sofia global siptrace on > > -BDF > On May 21, 2013 7:49 AM, wrote: > >> Tks for your reply, >> >> now i dont have error on log but, dtmf is not send again. >> >> please i need quick help >> >> tks advance >> >> Le 2013-05-20 23:11, Brian Foster a ?crit?: >> > 2013-05-17?09 >> [1]:39:54.214928?[WARNING]?mod_sofia.c:1363?Pass?2833?mode >> > may not work on a transcoded call. >> > >> > You shouldnt be transcoding if you can help it. Now, Im not sure >> if >> > that is an empty threat but you should enable late codec >> negotiation. >> > Information can be found here: >> > http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >> > >> > -BDF >> > On May 17, 2013 3:48 AM, wrote: >> > >> >> im so stupid :) >> >> tks >> >> >> >> http://pastebin.freeswitch.org/20933 [4] [1] >> >> >> >> called num is 022206... and when i try to use dtmf touch 5 its >> not >> >> works. >> >> >> >> tks >> >> >> >> Le 2013-05-17 09:25, Ken Rice a ?crit?: >> >> > it tells you the password in the popup... this is an anti spam >> >> thing >> >> > >> >> > KenSent from my iPad >> >> > >> >> > On May 17, 2013, at 0:46, Hermouet Erwan > [5] >> >> [2] [14]> >> >> > wrote: >> >> > >> >> >> On login i try my email...but don t work...i loose here >> >> >> >> >> >> Michael Collins a ?crit : >> >> >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >> >>> wrote: >> >> >>> >> >> >>>> I have the log but i never found how works pastebin ?? do >> you >> >> >>>> have tutorial ? >> >> >>> >> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> >> text >> >> >>> box, select FreeSWITCH Log as the syntax highlighting and >> then >> >> >>> click Send. Copy the URL from the browse address bar. it >> will >> >> be >> >> >>> something like: >> >> >>> http://pastebin.freeswitch.org/20927 [8] [5] [2] >> >> >>> >> >> >>> -MC >> >> >> >> >> >> Hermouet Erwan >> >> >> Responsable technique >> >> >> Bluetel >> >> > >> >> >> >> >> > >> >> > >> >> >> > >> > >> > > _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org [9] [6] [4] >> >> >> http://www.freeswitchsolutions.com [10] [7] [5] >> >> >> >> >> >> >> >> >> [11] [8] [6] >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org [12] [9] [7] >> >> >> http://wiki.freeswitch.org [13] [10] [8] >> >> >> http://www.cluecon.com [14] [11] [9] >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org [15] [12] [10] >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [16] >> >> [13] [11] >> >> >> >> >> > >> >> > >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [17] >> >> [14] >> >> >> [12] >> >> >> http://www.freeswitch.org [18] [15] [13] >> >> > >> >> > >> >> > Links: >> >> > ------ >> >> > [1] mailto:ehermouet at bluetel.fr [19] [16] >> >> > [2] http://pastebin.freeswitch.org/20927 [20] [17] >> >> > [3] mailto:msc at freeswitch.org [21] [18] >> >> > [4] mailto:consulting at freeswitch.org [22] [19] >> >> > [5] http://www.freeswitchsolutions.com [23] [20] >> >> > [6] [24] [21] >> >> > [7] http://www.freeswitch.org [25] [22] >> >> > [8] http://wiki.freeswitch.org [26] [23] >> >> > [9] http://www.cluecon.com [27] [24] >> >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [28] [25] >> >> > [11] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [29] [26] >> >> > [12] >> http://lists.freeswitch.org/mailman/options/freeswitch-users [30] >> >> [27] >> >> > [13] http://www.freeswitch.org [31] [28] >> >> > [14] mailto:ehermouet at bluetel.fr [32] [29] >> >> >> >> >> > >> > >> > > _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [33] [30] >> >> http://www.freeswitchsolutions.com [34] [31] >> >> >> >> >> >> [35] [32] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [36] [33] >> >> http://wiki.freeswitch.org [37] [34] >> >> http://www.cluecon.com [38] [35] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [39] [36] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [40] [37] >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [41] >> >> [38] >> >> http://www.freeswitch.org [42] [39] >> > >> > >> > Links: >> > ------ >> > [1] http://pastebin.freeswitch.org/20933 [43] >> > [2] mailto:ehermouet at bluetel.fr [44] >> > [3] mailto:msc at freeswitch.org [45] >> > [4] mailto:ehermouet at bluetel.fr [46] >> > [5] http://pastebin.freeswitch.org/20927 [47] >> > [6] mailto:consulting at freeswitch.org [48] >> > [7] http://www.freeswitchsolutions.com [49] >> > [8] [50] >> > [9] http://www.freeswitch.org [51] >> > [10] http://wiki.freeswitch.org [52] >> > [11] http://www.cluecon.com [53] >> > [12] mailto:FreeSWITCH-users at lists.freeswitch.org [54] >> > [13] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [55] >> > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [56] >> > [15] http://www.freeswitch.org [57] >> > [16] mailto:ehermouet at bluetel.fr [58] >> > [17] http://pastebin.freeswitch.org/20927 [59] >> > [18] mailto:msc at freeswitch.org [60] >> > [19] mailto:consulting at freeswitch.org [61] >> > [20] http://www.freeswitchsolutions.com [62] >> > [21] [63] >> > [22] http://www.freeswitch.org [64] >> > [23] http://wiki.freeswitch.org [65] >> > [24] http://www.cluecon.com [66] >> > [25] mailto:FreeSWITCH-users at lists.freeswitch.org [67] >> > [26] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [68] >> > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [69] >> > [28] http://www.freeswitch.org [70] >> > [29] mailto:ehermouet at bluetel.fr [71] >> > [30] mailto:consulting at freeswitch.org [72] >> > [31] http://www.freeswitchsolutions.com [73] >> > [32] [74] >> > [33] http://www.freeswitch.org [75] >> > [34] http://wiki.freeswitch.org [76] >> > [35] http://www.cluecon.com [77] >> > [36] mailto:FreeSWITCH-users at lists.freeswitch.org [78] >> > [37] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [79] >> > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [80] >> > [39] http://www.freeswitch.org [81] >> > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation [82] >> > [41] mailto:ehermouet at bluetel.fr [83] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [84] >> http://www.freeswitchsolutions.com [85] >> >> >> [86] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [87] >> http://wiki.freeswitch.org [88] >> http://www.cluecon.com [89] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [90] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [91] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [92] >> http://www.freeswitch.org [93] > > > Links: > ------ > [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 > [2] http://wiki.freeswitch.org/wiki/Codec_negotiation > [3] mailto:ehermouet at bluetel.fr > [4] http://pastebin.freeswitch.org/20933 > [5] mailto:ehermouet at bluetel.fr > [6] mailto:msc at freeswitch.org > [7] mailto:ehermouet at bluetel.fr > [8] http://pastebin.freeswitch.org/20927 > [9] mailto:consulting at freeswitch.org > [10] http://www.freeswitchsolutions.com > [11] > [12] http://www.freeswitch.org > [13] http://wiki.freeswitch.org > [14] http://www.cluecon.com > [15] mailto:FreeSWITCH-users at lists.freeswitch.org > [16] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [17] http://lists.freeswitch.org/mailman/options/freeswitch-users > [18] http://www.freeswitch.org > [19] mailto:ehermouet at bluetel.fr > [20] http://pastebin.freeswitch.org/20927 > [21] mailto:msc at freeswitch.org > [22] mailto:consulting at freeswitch.org > [23] http://www.freeswitchsolutions.com > [24] > [25] http://www.freeswitch.org > [26] http://wiki.freeswitch.org > [27] http://www.cluecon.com > [28] mailto:FreeSWITCH-users at lists.freeswitch.org > [29] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [30] http://lists.freeswitch.org/mailman/options/freeswitch-users > [31] http://www.freeswitch.org > [32] mailto:ehermouet at bluetel.fr > [33] mailto:consulting at freeswitch.org > [34] http://www.freeswitchsolutions.com > [35] > [36] http://www.freeswitch.org > [37] http://wiki.freeswitch.org > [38] http://www.cluecon.com > [39] mailto:FreeSWITCH-users at lists.freeswitch.org > [40] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [41] http://lists.freeswitch.org/mailman/options/freeswitch-users > [42] http://www.freeswitch.org > [43] http://pastebin.freeswitch.org/20933 > [44] mailto:ehermouet at bluetel.fr > [45] mailto:msc at freeswitch.org > [46] mailto:ehermouet at bluetel.fr > [47] http://pastebin.freeswitch.org/20927 > [48] mailto:consulting at freeswitch.org > [49] http://www.freeswitchsolutions.com > [50] > [51] http://www.freeswitch.org > [52] http://wiki.freeswitch.org > [53] http://www.cluecon.com > [54] mailto:FreeSWITCH-users at lists.freeswitch.org > [55] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [56] http://lists.freeswitch.org/mailman/options/freeswitch-users > [57] http://www.freeswitch.org > [58] mailto:ehermouet at bluetel.fr > [59] http://pastebin.freeswitch.org/20927 > [60] mailto:msc at freeswitch.org > [61] mailto:consulting at freeswitch.org > [62] http://www.freeswitchsolutions.com > [63] > [64] http://www.freeswitch.org > [65] http://wiki.freeswitch.org > [66] http://www.cluecon.com > [67] mailto:FreeSWITCH-users at lists.freeswitch.org > [68] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [69] http://lists.freeswitch.org/mailman/options/freeswitch-users > [70] http://www.freeswitch.org > [71] mailto:ehermouet at bluetel.fr > [72] mailto:consulting at freeswitch.org > [73] http://www.freeswitchsolutions.com > [74] > [75] http://www.freeswitch.org > [76] http://wiki.freeswitch.org > [77] http://www.cluecon.com > [78] mailto:FreeSWITCH-users at lists.freeswitch.org > [79] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [80] http://lists.freeswitch.org/mailman/options/freeswitch-users > [81] http://www.freeswitch.org > [82] http://wiki.freeswitch.org/wiki/Codec_negotiation > [83] mailto:ehermouet at bluetel.fr > [84] mailto:consulting at freeswitch.org > [85] http://www.freeswitchsolutions.com > [86] > [87] http://www.freeswitch.org > [88] http://wiki.freeswitch.org > [89] http://www.cluecon.com > [90] mailto:FreeSWITCH-users at lists.freeswitch.org > [91] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [92] http://lists.freeswitch.org/mailman/options/freeswitch-users > [93] http://www.freeswitch.org > [94] mailto:ehermouet at bluetel.fr From a.mykhalkiv at kwebbl.com Tue May 21 11:12:33 2013 From: a.mykhalkiv at kwebbl.com (Anatolii) Date: Tue, 21 May 2013 00:12:33 -0700 (PDT) Subject: [Freeswitch-users] Problem with enable-post-var Message-ID: <1369120353358-7590893.post@n2.nabble.com> Hello all. In mod_xml_curl in binding configuration I try make dynamic "gateway-url". For example, $${my_var}*" bindings="configuration"/> Also try to add 'enable-post-var' -- didn't work too. Please, help anyone. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-enable-post-var-tp7590893.html Sent from the freeswitch-users mailing list archive at Nabble.com. From a.mykhalkiv at kwebbl.com Tue May 21 11:58:11 2013 From: a.mykhalkiv at kwebbl.com (Anatolii) Date: Tue, 21 May 2013 00:58:11 -0700 (PDT) Subject: [Freeswitch-users] Problem with enable-post-var In-Reply-To: <1369120353358-7590893.post@n2.nabble.com> References: <1369120353358-7590893.post@n2.nabble.com> Message-ID: <1369123091086-7590894.post@n2.nabble.com> I have found solution: *< param name="use-dynamic-url" value="yes"/>* < param name="gateway-url" value="http://example.com/freeswitch/configuration/${conf_name}" bindings="configuration"/> In vars.xml (variable 'conf_name' is empty default) And in dialplan < action application='set_global' data='conf_name=test at test'/> Set conf_name some value or make it empty (data='conf_name=') -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-enable-post-var-tp7590893p7590894.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gmangudai at gmail.com Tue May 21 12:13:29 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Tue, 21 May 2013 16:13:29 +0800 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference Message-ID: Dear Support, my question for using freeswitch is: how to turn an ongoing call in to a conference so i can have the third party in without breaking up the call (no hangup, dial stuff)? something like call intrusion?A and B are already talking on the phone, then C wants to talk to both A and B. any response will be greatly appriciated. Jacob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/fd7c2de6/attachment.html From bdfoster at davri.com Tue May 21 16:59:33 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 08:59:33 -0400 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference In-Reply-To: References: Message-ID: mod_spy or have your client manage a 3 way call. Please explain your situation a little more we might be able to give you a better answer. Is caller C inbound or outbound to the call? On May 21, 2013 8:55 AM, "Vincent Xia" wrote: > Dear Support, > > my question for using freeswitch is: > > how to turn an ongoing call in to a conference so i can have the third > party in without breaking up the call (no hangup, dial stuff)? > something like call intrusion?A and B are already talking on the phone, > then C wants to talk to both A and B. > > any response will be greatly appriciated. > > Jacob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/8ac760c4/attachment.html From cmrienzo at gmail.com Tue May 21 16:59:42 2013 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 21 May 2013 08:59:42 -0400 Subject: [Freeswitch-users] mod_unimrcp dont send invite while using a Mrcp v2 profile In-Reply-To: References: Message-ID: I look at it tonight. On Mon, May 20, 2013 at 9:33 PM, Leonardo P. Bidinoto < leonardo.bidinoto at voicetechnology.com.br> wrote: > Hi All, > Im new at using the mod_unimrcp. > I'm having some troubles when configuring to using several profiles at > same time. > I have configured 3 profiles(2 with Nuance and 1 Verbio, all using MRCPv2). > But when i try to use a TTS or ASR from any of those profile, FS channel > get stuck and i get a warning message telling that the time for the MRCP > server to respond have expired. > I have used wireshark to capture the packages, and when i call speak or > detect_speech function of FS, i dont see any SIP INVITE going from FS to > the MRCP Server. > > Did i miss something at configuring profiles? Did anyone get something > similar to this situation? I'm using the examples from FS mod_unimrcp page. > > Any tip will help a lot. > > Thanks > > -- > Leonardo Pires Bidinoto > Voice Technology > www.voicetechnology.com.br > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/de1af246/attachment-0001.html From jason.holden at start.ca Tue May 21 17:40:56 2013 From: jason.holden at start.ca (Jason Holden) Date: Tue, 21 May 2013 09:40:56 -0400 Subject: [Freeswitch-users] answer calls during opening message of fifo Message-ID: <54D7102E22A84148862A9BDCFE7CAE1F@bob> I would lke a welcome message to play when a caller connects to FIFO where the agents are able to answer their phone while this message is played. Currently I am using play sound but it does not ring the agent phones during this messages since fifo has been entered. Does anyone have any recommendations? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/dd5899be/attachment.html From andrew at cassidywebservices.co.uk Tue May 21 17:51:22 2013 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 21 May 2013 14:51:22 +0100 Subject: [Freeswitch-users] Google Talk XMPP Support Message-ID: I've heard rumors that XMPP support in Google Talk is being killed. I'm guessing this means no more Google Talk in FreeSWITCH? -- *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/20130521/af83198b/attachment.html From nneul at mst.edu Tue May 21 18:02:17 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Tue, 21 May 2013 09:02:17 -0500 Subject: [Freeswitch-users] Google Talk XMPP Support In-Reply-To: References: Message-ID: <519B7E69.4070506@mst.edu> It's unclear to me (most articles are pretty vague) as to whether "XMPP" is being killed or "XMPP Federation". It sounds like federation is the current target, but can't really confirm it. If so, that may not impact regular xmpp client usage such as pidgin/freeswitch/etc. -- Nathan On 05/21/2013 08:51 AM, Andrew Cassidy wrote: > I've heard rumors that XMPP support in Google Talk is being killed. I'm guessing this means no more Google Talk in > FreeSWITCH? > > -- > *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 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From ml-ktk at netlabs.org Tue May 21 18:48:38 2013 From: ml-ktk at netlabs.org (Adrian Gschwend) Date: Tue, 21 May 2013 16:48:38 +0200 Subject: [Freeswitch-users] Google Talk XMPP Support In-Reply-To: References: Message-ID: <519B8946.2010606@netlabs.org> On 21.05.13 15:51, Andrew Cassidy wrote: > I've heard rumors that XMPP support in Google Talk is being killed. I'm > guessing this means no more Google Talk in FreeSWITCH? found some more information here, still not too many details though http://windowspbx.blogspot.ch/2013/05/hangouts-wont-hangout-with-other.html regards Adrian From philippe at ppmt.org Tue May 21 21:40:07 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 21 May 2013 13:40:07 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: For the inbound I didn't comment because I can't understand how your dialplan would link your gateway to your extension On 21 May 2013 00:50, Mike Hendrie wrote: > I will take a closer look at the gateways. However, your email looks like > it was sent before you completed it. > > What about the inbound DID? Any ideas? > > > On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: > >> I am no expert so I could be wrong but: >> >> First your outbound is no registered (NOREG) so that is most likely going >> to prevent outgoing call >> >> Also you refer to default_gateway in your dialplan....what value is it >> set to? Check vars.xmls >> >> I normally use directly the name of my gateway instead of using variable >> (might not be the best way though ) >> >> >> >> But you have the >> On 13-05-19 11:39 PM, Mike Hendrie wrote: >> >> Thank you for your response. Per your request here is the result of the sofia >> status command: >> >> freeswitch at internal> sofia status >> Name Type >> Data State >> >> ================================================================================================= >> 10.1.1.5 alias >> internal ALIASED >> internal profile >> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >> external profile >> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >> external::example.com gateway >> sip:joeuser at example.com NOREG >> external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED >> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >> NOREG >> GothamCity.xom alias >> internal ALIASED >> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >> RUNNING (0) >> >> ================================================================================================= >> 3 profiles 2 aliases >> >> >> ========================================== >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> >> >> >> >> ========================================== >> >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> ========================================== >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/07ba0206/attachment-0001.html From clive18 at webmail.co.za Tue May 21 22:09:56 2013 From: clive18 at webmail.co.za (clive engelberg) Date: Tue, 21 May 2013 20:09:56 +0200 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy In-Reply-To: References: Message-ID: Hi You can get help in exchange for some netflix movies:) jk :) Freeswitch will act as a stateful proxy, meaning it will remain in control of the call, even though RTP goes point to point. Why would you want to be able to shut down FS anyways? Clive On Mon, 20 May 2013 19:10:09 -0700 Oleg Stolyar wrote Hi, ? I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below which simply sets bypass_media to true and then bridges to another FreeSWITCH server. ? However, when during the call I shut down the proxy FS, the call is immediately dropped. Why is that? Is there a way to keep it going? I understand that in this case I won't be able to properly send the BYE signal when one of the parties hangs up and that's OK. ? I tried using redirect and deflect instead of the bridge but those don't seem to work at all - probably because my UAs don't know how to handle redirects. ? ? Links: ------ [1] mailto:1000 at 10.2.182.243 ____________________________________________________________ South Africas premier free email service - www.webmail.co.za For super low premiums, click here. http://www.dialdirect.co.za/smart-can-get-a-parrot-hands-free-kit?vdn=15752 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/78a7738f/attachment.html From mike at jerris.com Tue May 21 22:18:27 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 21 May 2013 14:18:27 -0400 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy In-Reply-To: References: Message-ID: <0D645108-3498-409D-ADD2-09D86CBBDF68@jerris.com> FreeSWITCH really isn't a proxy, its a B2BUA. That being said, if you stop freeswitch with "fsctl shutdown now" it will exit without cleaning up any running calls. On May 21, 2013, at 2:09 PM, clive engelberg wrote: > Hi > > You can get help in exchange for some netflix movies:) > > jk :) > > Freeswitch will act as a stateful proxy, meaning it will remain in control of the call, even though RTP goes point to point. > > Why would you want to be able to shut down FS anyways? > > Clive > > On Mon, 20 May 2013 19:10:09 -0700 Oleg Stolyar wrote > > Hi, > > I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below which simply sets bypass_media to true and then bridges to another FreeSWITCH server. > > However, when during the call I shut down the proxy FS, the call is immediately dropped. Why is that? Is there a way to keep it going? I understand that in this case I won't be able to properly send the BYE signal when one of the parties hangs up and that's OK. > > I tried using redirect and deflect instead of the bridge but those don't seem to work at all - probably because my UAs don't know how to handle redirects. > > > > > > > > > > South Africa premier free email service - webmail.co.za > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/260e5bfb/attachment.html From bdfoster at davri.com Tue May 21 22:21:44 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 14:21:44 -0400 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy In-Reply-To: References: Message-ID: I'd almost be looking at something like OpenSIPS as FS is a back2back UA, not a proxy. On May 21, 2013 2:17 PM, "clive engelberg" wrote: > Hi > > You can get help in exchange for some netflix movies:) > > jk :) > > Freeswitch will act as a stateful proxy, meaning it will remain in control > of the call, even though RTP goes point to point. > > Why would you want to be able to shut down FS anyways? > > Clive > > On Mon, 20 May 2013 19:10:09 -0700 Oleg Stolyar > wrote > > Hi, > > I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below > which simply sets bypass_media to true and then bridges to another > FreeSWITCH server. > > However, when during the call I shut down the proxy FS, the call is > immediately dropped. Why is that? Is there a way to keep it going? I > understand that in this case I won't be able to properly send the BYE > signal when one of the parties hangs up and that's OK. > > I tried using redirect and deflect instead of the bridge but those don't > seem to work at all - probably because my UAs don't know how to handle > redirects. > > > > > > > > > > ------------------------------ > South Africa premier free email service - webmail.co.za > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/ed80b6ea/attachment.html From anthony.minessale at gmail.com Tue May 21 22:29:26 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 21 May 2013 13:29:26 -0500 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference In-Reply-To: References: Message-ID: The three_way lets you attach to an existing uuid, like eavesdrop only with the audio path full open in all directions. On Tue, May 21, 2013 at 7:59 AM, Brian Foster wrote: > mod_spy or have your client manage a 3 way call. Please explain your > situation a little more we might be able to give you a better answer. Is > caller C inbound or outbound to the call? > On May 21, 2013 8:55 AM, "Vincent Xia" wrote: > >> Dear Support, >> >> my question for using freeswitch is: >> >> how to turn an ongoing call in to a conference so i can have the third >> party in without breaking up the call (no hangup, dial stuff)? >> something like call intrusion?A and B are already talking on the phone, >> then C wants to talk to both A and B. >> >> any response will be greatly appriciated. >> >> Jacob >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/90e156bb/attachment-0001.html From jalsot at gmail.com Tue May 21 23:53:26 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Tue, 21 May 2013 21:53:26 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips Message-ID: Hello, Do you have any recommendations regarding how to set up correctly (for production) CentOS5 openvz and FS 1.2.stable? Is there any trick to tuneup the system to be rock solid? Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that something is not well configured around timers, however mod_posix_timer did not help anything (running FS with -rp). We use our own bare metal and can reproduce those delatas eirher when only one VE is on the HW. Maybe time to check out centos6 with openvz? Any idea, recommendation, experience can be very helpful. Regards, Jalsot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/b70b0f5d/attachment.html From cesar.bermudez at gmail.com Wed May 22 00:23:09 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 21 May 2013 14:23:09 -0600 Subject: [Freeswitch-users] files.freeswitch.org down Message-ID: hi all, is the domain files.freeswitch.org down? i cannot download the finish the install ( the sounds ) or its just me? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/4b4ff703/attachment.html From krice at freeswitch.org Wed May 22 00:39:13 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 21 May 2013 15:39:13 -0500 Subject: [Freeswitch-users] files.freeswitch.org down In-Reply-To: Message-ID: It was having issues its back On 5/21/13 3:23 PM, "Cesar Bermudez" wrote: > hi all, is the domain files.freeswitch.org > down? i cannot download the finish the install ( the sounds ) > or its just me? > > Regards. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/7a235820/attachment.html From sos at sokhapkin.dyndns.org Wed May 22 00:40:46 2013 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 21 May 2013 16:40:46 -0400 Subject: [Freeswitch-users] files.freeswitch.org down In-Reply-To: References: Message-ID: <6722812.4hHn6AvMPO@sos> Works OK to me. On Tuesday 21 May 2013 14:23:09 Cesar Bermudez wrote: > hi all, is the domain files.freeswitch.org down? i cannot download the > finish the install ( the sounds ) > or its just me? > > Regards. From schoch+freeswitch.org at xwin32.com Wed May 22 00:43:53 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 21 May 2013 13:43:53 -0700 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: I had the opportunity to capture the data to/from the phone. I used this packet capture command, on the FS host itself: # /usr/sbin/tcpdump -p -s 0 -w jon.pcap -i eth1 host 192.168.4.235 10169 packets captured 10169 packets received by filter 0 packets dropped by kernel The number dialed was 18002729872, the IRS EFTPS IVR. The first voice prompt is to enter the 9-digit taxpayer identification number. In Wireshark, I can clearly see the numbers being entered, as RFC2833 RTP DTMF events. One thing I notice is that I get several RTP packets for each digit. (I assume that's normal, but I'm new at this.) For example, here is one digit: 6507 87.427666 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 6508 87.431287 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14414, Time=499365981 6509 87.447662 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 6510 87.451285 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14415, Time=499366141 6511 87.467655 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 6512 87.471284 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14416, Time=499366301 6513 87.487656 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 6514 87.491286 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14417, Time=499366461 6515 87.507665 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 6516 87.511285 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14418, Time=499366621 6517 87.527677 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 (end) 6518 87.531284 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14419, Time=499366781 6519 87.547688 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 (end) 6520 87.551336 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x71C6FAAF, Seq=14420, Time=499366941 6521 87.567658 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP Event, DTMF Two 2 (end) This sequence happens 9 times with different numbers, as the 9-digit TIN is entered. At 80.5 seconds (according to the Wireshark player), the called party says: "The tax identification number must be nine digits. Please re-enter." The last digit event was at timestamp 2495866430 (phone) or 499404541 (FS). I don't know how to convert RTP timestamps to actual time. What's the next step to diagnosing the problem? -- Steve On Thu, May 9, 2013 at 1:14 PM, Michael Collins wrote: > Are you sending in-band DTMFs or RFC2833? Or both? Get a pcap of the call, > including the RTP, and analyze it in Wireshark. See what you are actually > sending. That means looking at the flow (to see if you send RTPEVENTS, i.e. > RFC2833 DTMFs) and also listening to the decoded audio with the player. > > -MC > > > On Thu, May 9, 2013 at 12:59 PM, Steven Schoch < > schoch+freeswitch.org at xwin32.com> wrote: > >> One of my users complains of many problems calling automated systems that >> take DTMF input. In this case it was an IRS payment entry system that would >> skip or duplicate button presses. >> >> Our setup is pretty generic: >> Polycom SoundPoint IP 320 SIP -> dedicated LAN -> Freeswitch (on CentOS, >> on XenServer) -> Comcast static IP -> Flowroute.com -> some IRS 800 number. >> >> Is this just the way it is, or is there something I can do to make it >> work better? >> >> -- >> Steve >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/3a57e2bc/attachment-0001.html From cesar.bermudez at gmail.com Wed May 22 01:04:50 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 21 May 2013 15:04:50 -0600 Subject: [Freeswitch-users] files.freeswitch.org down In-Reply-To: References: Message-ID: Thx On Tue, May 21, 2013 at 2:39 PM, Ken Rice wrote: > It was having issues its back > > > On 5/21/13 3:23 PM, "Cesar Bermudez" wrote: > > hi all, is the domain files.freeswitch.org > down? i cannot download the finish the install ( the sounds ) > > or its just me? > > Regards. > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/58f83a19/attachment.html From jrlh at outlook.com Tue May 21 20:11:31 2013 From: jrlh at outlook.com (JohnGalt1717) Date: Tue, 21 May 2013 09:11:31 -0700 (PDT) Subject: [Freeswitch-users] Codec Negotiation Error Lync/UCMA -> external provider Message-ID: <1369152691371-7590911.post@n2.nabble.com> Hi there, I'm a newbee, but I've been reading everything I can and I can't find anything that assists with this. I have a UCMA app that is internal that communicates on the internal sip port (set to 5042) and it bridges with dialplan to the external sip trunk (trying 2, one of which is siproutes). If I enable proxy, the call rings out and I get dead air on the other side, but it won't ever properly establish the call in the UCMA app because of some issue with codecs from what I can tell. Obviously I don't want to use proxy mode, what I really want to do is use it straight up and have FS do the transcoding etc. The problem is that once I do this I get a Codec Negotiation error and I can't figure out why. Anyone have any ideas as to why it's freaking out like this? (log below) Thanks! 2013-05-21 12:09:41.570884 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1369129003 1369129004 IN IP4 10.8.0.158 s=FreeSWITCH c=IN IP4 10.8.0.158 t=0 0 m=audio 23578 RTP/AVP 9 0 8 101 13 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:114 (sofia/external/+16032897972) State Change CS_INIT -> CS_ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+16032897972) State INIT going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+16032897972) Running State Change CS_ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+16032897972) State ROUTING 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:137 sofia/external/+16032897972 SOFIA ROUTING 2013-05-21 12:09:41.570884 [DEBUG] switch_ivr_originate.c:67 (sofia/external/+16032897972) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 (sofia/external/+16032897972) State ROUTING going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+16032897972) Running State Change CS_CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 (sofia/external/+16032897972) State CONSUME_MEDIA 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 (sofia/external/+16032897972) State CONSUME_MEDIA going to sleep 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:41.570884 [DEBUG] sofia.c:5742 Channel sofia/external/+16032897972 entering state [calling][0] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal sofia/external/+16032897972 [BREAK] 2013-05-21 12:09:43.891308 [INFO] sofia.c:966 sofia/external/+16032897972 Update Callee ID to "+16032897972" 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5742 Channel sofia/external/+16032897972 entering state [proceeding][183] 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5751 Remote SDP: v=0 o=Sonus_UAC 10608 14579 IN IP4 207.239.33.59 s=SIP Media Capabilities c=IN IP4 207.239.33.59 t=0 0 m=audio 10460 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=maxptime:20 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:1813 Set Codec sofia/external/+16032897972 PCMU/8000 20 ms 160 samples 64000 bits 2013-05-21 12:09:43.891308 [DEBUG] switch_core_codec.c:111 sofia/external/+16032897972 Original read codec set to PCMU:0 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2876 Set 2833 dtmf send payload to 101 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:3800 AUDIO RTP [sofia/external/+16032897972] 10.8.0.158 port 23578 -> 207.239.33.59 port 10460 codec: 0 m s: 20 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:2659 Starting timer [soft] 160 bytes per 20ms 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: START SESSION INITIALIZATION. sID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ZID=306130383030396530393363. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Loading User's profile: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: allowclear: OFF 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: autosecure: ON 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: disclose_bit: OFF 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: signal. role: Initiator 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: TTL: 4294967295 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: SAS schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 B256 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 B32 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Ciphers: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 AES3 2013-05-21 12:09:4 3.891308 [DEBUG] switch_rtp.c:1231 AES1 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: PK schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 EC25 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 DH3k 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 DH2k 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 Mult 201 3-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATL: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 HS32 2013-05-21 12:09: 43.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Hashes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 S256 2013-05-21 12:09:4 3.891308 [DEBUG] switch_rtp.c:1231 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Session initialization - DONE. sID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATTACH NEW STREAM to sID=4: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: Stream ID=0 UNKNOWN switching ---> . 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Empty slot was found - initializing new stream with ID=4. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: Preparing ZRTP Hello according to the Session profile. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: ATTACH NEW STREAM - DONE. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp engine]: START STREAM ID=4 mode=CLEAR state=ACTIVE. 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: Stream ID=4 CLEAR switching ---> . 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp utils]: Send ssrc=1428962591 seq=54611 size=144. Stream 4:CLEAR:START 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4145 Set 2833 dtmf send payload to 101 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4151 Set 2833 dtmf receive payload to 101 2013-05-21 12:09:43.891308 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/+16032897972! 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3273 Send signal sofia/internal/+16032897972 at 173.255.174.20 [BREAK] 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3277 (sofia/external/+16032897972) Callstate Change DOWN -> EARLY 2013-05-21 12:09:43.911297 [DEBUG] switch_ivr_originate.c:411 Setting codec string on sofia/internal/+16032897972 at 173.255.174.20 to PCMU at 8000h@20i 2013-05-21 12:09:43.911297 [INFO] switch_ivr_originate.c:3416 Sending early media 2013-05-21 12:09:43.911297 [ERR] switch_core_media.c:2549 a=crypto in RTP/AVP, refer to rfc3711 2013-05-21 12:09:43.911297 [ERR] mod_sofia.c:2081 CODEC NEGOTIATION ERROR. SDP: v=0 o=- 1 0 IN IP4 10.8.0.158 s=session c=IN IP4 10.8.0.158 b=CT:10000 t=0 0 m=audio 2908 RTP/AVP 9 112 111 0 8 116 13 118 97 101 c=IN IP4 10.8.0.158 a=rtpmap:9 G722/8000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:116 AAL2-G726-32/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16,36 a=rtcp:2909 a=label:main-audio a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BqtEfRv/sMasd3GS0I4FJRX5hh7BKVt+r8/vKbOk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Codec-Negotiation-Error-Lync-UCMA-external-provider-tp7590911.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at hendrienet.com Tue May 21 21:47:57 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Tue, 21 May 2013 12:47:57 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: As you may have guessed, I'm new to the application. Can you say how I should configure the inbound? Mike On May 21, 2013 12:43 PM, "Philippe Le Toquin" wrote: > For the inbound I didn't comment because I can't understand how your > dialplan would link your gateway to your extension > > > > > On 21 May 2013 00:50, Mike Hendrie wrote: > >> I will take a closer look at the gateways. However, your email looks >> like it was sent before you completed it. >> >> What about the inbound DID? Any ideas? >> >> >> On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: >> >>> I am no expert so I could be wrong but: >>> >>> First your outbound is no registered (NOREG) so that is most likely >>> going to prevent outgoing call >>> >>> Also you refer to default_gateway in your dialplan....what value is it >>> set to? Check vars.xmls >>> >>> I normally use directly the name of my gateway instead of using variable >>> (might not be the best way though ) >>> >>> >>> >>> But you have the >>> On 13-05-19 11:39 PM, Mike Hendrie wrote: >>> >>> Thank you for your response. Per your request here is the result of the sofia >>> status command: >>> >>> freeswitch at internal> sofia status >>> Name Type >>> Data State >>> >>> ================================================================================================= >>> 10.1.1.5 alias >>> internal ALIASED >>> internal profile >>> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >>> external profile >>> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >>> external::example.com gateway >>> sip:joeuser at example.com NOREG >>> external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED >>> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >>> NOREG >>> GothamCity.xom alias >>> internal ALIASED >>> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >>> RUNNING (0) >>> >>> ================================================================================================= >>> 3 profiles 2 aliases >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ========================================== >>> >>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/$1"/> >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/$1"/> >>> >>> >>> >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${outbound_caller_name}"/> >>> >> value="$${outbound_caller_id}"/> >>> >>> >>> >>> >>> >>> ========================================== >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/948e666c/attachment-0001.html From mike at hendrienet.com Wed May 22 00:20:57 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Tue, 21 May 2013 15:20:57 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: <519A9A81.7020609@ppmt.org> References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: I am looking in the vars.xml file and am not seeing any reference to a gateway: ====== vars.xml On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: > I am no expert so I could be wrong but: > > First your outbound is no registered (NOREG) so that is most likely going > to prevent outgoing call > > Also you refer to default_gateway in your dialplan....what value is it set > to? Check vars.xmls > > I normally use directly the name of my gateway instead of using variable > (might not be the best way though ) > > > > But you have the > On 13-05-19 11:39 PM, Mike Hendrie wrote: > > Thank you for your response. Per your request here is the result of the sofia > status command: > > freeswitch at internal> sofia status > Name Type > Data State > > ================================================================================================= > 10.1.1.5 alias > internal ALIASED > internal profile > sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) > external profile > sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom > REGED > external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom > NOREG > GothamCity.xom alias > internal ALIASED > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > > ================================================================================================= > 3 profiles 2 aliases > > > ========================================== > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > > > > ========================================== > > /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > > data="effective_caller_id_number=${outbound_caller_id_number}"/> > data="effective_caller_id_name=${outbound_caller_id_name}"/> > data="sofia/gateway/${default_gateway}/$1"/> > > > > > > ========================================== > /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ========================================== > /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > ========================================== > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130521/76415e07/attachment-0001.html From mike at hendrienet.com Wed May 22 00:23:06 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Tue, 21 May 2013 15:23:06 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: I think I am missing the basic understanding of how the DID connects to the gateway then to the extension. Which file does what. thanks!! On Tue, May 21, 2013 at 3:20 PM, Mike Hendrie wrote: > I am looking in the vars.xml file and am not seeing any reference to a > gateway: > > ====== > vars.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> > > > data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> > > > > > > > > > > data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> > > data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> > > data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: > >> I am no expert so I could be wrong but: >> >> First your outbound is no registered (NOREG) so that is most likely going >> to prevent outgoing call >> >> Also you refer to default_gateway in your dialplan....what value is it >> set to? Check vars.xmls >> >> I normally use directly the name of my gateway instead of using variable >> (might not be the best way though ) >> >> >> >> But you have the >> On 13-05-19 11:39 PM, Mike Hendrie wrote: >> >> Thank you for your response. Per your request here is the result of the sofia >> status command: >> >> freeswitch at internal> sofia status >> Name Type >> Data State >> >> ================================================================================================= >> 10.1.1.5 alias >> internal ALIASED >> internal profile >> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >> external profile >> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >> external::example.com gateway >> sip:joeuser at example.com NOREG >> external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED >> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >> NOREG >> GothamCity.xom alias >> internal ALIASED >> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >> RUNNING (0) >> >> ================================================================================================= >> 3 profiles 2 aliases >> >> >> ========================================== >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> >> >> >> >> ========================================== >> >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> >> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> ========================================== >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130521/96957f07/attachment-0001.html From msc at freeswitch.org Wed May 22 01:26:44 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 14:26:44 -0700 Subject: [Freeswitch-users] Notification of an event in the dialplan In-Reply-To: References: Message-ID: Yeah, this should work. OTTOMH I'd say do this: Let us know how that works... -MC On Mon, May 20, 2013 at 10:40 PM, Avi Marcus wrote: > Hmm, this looks promising: > http://wiki.freeswitch.org/wiki/Mod_commands#bg_system > Execute a system command in the background. > > Usage: bg_system > > Please update the wiki if you get more information about how that works... > > -Avi > > On Tue, May 21, 2013 at 2:22 AM, Christopher Aloi wrote: > >> I was thinking about that - my concern was "and waits for the result". >> Could that potentially hang call processing? If I pass to a python -> >> sendmail script and wait for the result, if my script hangs, I would have >> these calls waiting for the result (right?) >> >> On Monday, May 20, 2013, Michael Collins wrote: >> >>> A simple approach might be to use the system app: >>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system >>> >>> If you have a simple shell script that launches an email that would a >>> quick and easy way to get going. >>> >>> -MC >>> >>> >>> On Mon, May 20, 2013 at 1:25 PM, Christopher Aloi wrote: >>> >>>> >>>> I'm currently using FreeSwitch between two systems as a "rate limiter' >>>> with the limit application. >>>> >>>> I bring calls in, run through a hash to identify a loop - if my hash >>>> variable is met, I transfer the call to a new extension where I log the >>>> event and return a 503 to the source. If not I return a 302 to the source >>>> to send the call on its way. >>>> >>>> I'd like a way to identify these events to our NOC via email or a SNMP >>>> trap but I am not sure the best way to do it. >>>> >>>> The alert would fall into the 503 extension. >>>> >>>> What is the best way to accomplish this? >>>> >>>> I'd like to keep the call flow as simple as possible. >>>> >>>> Any ideas? >>>> >>>> Thanks - >>>> >>>> - Chris >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/c21712ba/attachment.html From msc at freeswitch.org Wed May 22 01:37:27 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 14:37:27 -0700 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Hi Mike, No worries - this can be frustrating at first. You'll get the hang of it shortly. Okay, technically speaking there is no explicit link between a gateway and an extension. Rather, the inbound call is handled by the dialplan. If you're using the vanilla example configs (which you should be to start out) then your inbound call is handled in the public dialplan context. This means that you need to have a file in conf/dialplan/public/ to handle your inbound call. It looks to me like your GothamCity.xom.xml file would work, but you'll need to make at least two changes: #1 - move the file from the conf/dialplan/default/ directory into conf/dialplan/public/ #2 - add the regular expression for your DID number. For example, if your DID number was 800-555-1212 you would do something like this: Hope this helps. -MC On Tue, May 21, 2013 at 1:23 PM, Mike Hendrie wrote: > I think I am missing the basic understanding of how the DID connects to > the gateway then to the extension. Which file does what. > > thanks!! > > > On Tue, May 21, 2013 at 3:20 PM, Mike Hendrie wrote: > >> I am looking in the vars.xml file and am not seeing any reference to a >> gateway: >> >> ====== >> vars.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> >> >> >> > data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> >> >> >> >> >> >> >> >> >> >> > data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> >> >> > data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> >> >> > data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: >> >>> I am no expert so I could be wrong but: >>> >>> First your outbound is no registered (NOREG) so that is most likely >>> going to prevent outgoing call >>> >>> Also you refer to default_gateway in your dialplan....what value is it >>> set to? Check vars.xmls >>> >>> I normally use directly the name of my gateway instead of using variable >>> (might not be the best way though ) >>> >>> >>> >>> But you have the >>> On 13-05-19 11:39 PM, Mike Hendrie wrote: >>> >>> Thank you for your response. Per your request here is the result of the sofia >>> status command: >>> >>> freeswitch at internal> sofia status >>> Name Type >>> Data State >>> >>> ================================================================================================= >>> 10.1.1.5 alias >>> internal ALIASED >>> internal profile >>> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >>> external profile >>> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >>> external::example.com gateway >>> sip:joeuser at example.com NOREG >>> external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED >>> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >>> NOREG >>> GothamCity.xom alias >>> internal ALIASED >>> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >>> RUNNING (0) >>> >>> ================================================================================================= >>> 3 profiles 2 aliases >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ========================================== >>> >>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/$1"/> >>> >>> >>> >>> >>> >>> >>> >> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> >> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> >> data="sofia/gateway/${default_gateway}/$1"/> >>> >>> >>> >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ========================================== >>> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${outbound_caller_name}"/> >>> >> value="$${outbound_caller_id}"/> >>> >>> >>> >>> >>> >>> ========================================== >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/cc5f8e1d/attachment-0001.html From msc at freeswitch.org Wed May 22 01:51:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 14:51:26 -0700 Subject: [Freeswitch-users] DTMF In-Reply-To: References: Message-ID: Okay, this is helpful. So you're definitely seeing RFC2833 digits from your telephone device. So the next question is what is going out to your provider. It just so happens that you didn't capture that leg of the call, but that's okay, you can just make another test call. Capture the packets heading out to Flowroute and see what's in there. I would listen to that capture in the Wireshark player and see if there are any digits going out. Also, yes, the multiple packets for a single RFC2833 digit is quite normal. -MC On Tue, May 21, 2013 at 1:43 PM, Steven Schoch < schoch+freeswitch.org at xwin32.com> wrote: > I had the opportunity to capture the data to/from the phone. I used this > packet capture command, on the FS host itself: > > # /usr/sbin/tcpdump -p -s 0 -w jon.pcap -i eth1 host 192.168.4.235 > 10169 packets captured > 10169 packets received by filter > 0 packets dropped by kernel > > The number dialed was 18002729872, the IRS EFTPS IVR. The first voice > prompt is to enter the 9-digit taxpayer identification number. > > In Wireshark, I can clearly see the numbers being entered, as RFC2833 RTP > DTMF events. One thing I notice is that I get several RTP packets for each > digit. (I assume that's normal, but I'm new at this.) For example, here is > one digit: > > 6507 87.427666 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 > 6508 87.431287 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14414, Time=499365981 > 6509 87.447662 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 > 6510 87.451285 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14415, Time=499366141 > 6511 87.467655 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 > 6512 87.471284 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14416, Time=499366301 > 6513 87.487656 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 > 6514 87.491286 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14417, Time=499366461 > 6515 87.507665 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 > 6516 87.511285 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14418, Time=499366621 > 6517 87.527677 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 (end) > 6518 87.531284 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14419, Time=499366781 > 6519 87.547688 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 (end) > 6520 87.551336 192.168.4.1 192.168.4.235 RTP 214 PT=ITU-T G.711 PCMU, > SSRC=0x71C6FAAF, Seq=14420, Time=499366941 > 6521 87.567658 192.168.4.235 192.168.4.1 RTP EVENT 60 Payload type=RTP > Event, DTMF Two 2 (end) > > This sequence happens 9 times with different numbers, as the 9-digit TIN > is entered. At 80.5 seconds (according to the Wireshark player), the called > party says: "The tax identification number must be nine digits. Please > re-enter." The last digit event was at timestamp 2495866430 (phone) > or 499404541 (FS). I don't know how to convert RTP timestamps to actual > time. > > What's the next step to diagnosing the problem? > > -- > Steve > > > On Thu, May 9, 2013 at 1:14 PM, Michael Collins wrote: > >> Are you sending in-band DTMFs or RFC2833? Or both? Get a pcap of the >> call, including the RTP, and analyze it in Wireshark. See what you are >> actually sending. That means looking at the flow (to see if you send >> RTPEVENTS, i.e. RFC2833 DTMFs) and also listening to the decoded audio with >> the player. >> >> -MC >> >> >> On Thu, May 9, 2013 at 12:59 PM, Steven Schoch < >> schoch+freeswitch.org at xwin32.com> wrote: >> >>> One of my users complains of many problems calling automated systems >>> that take DTMF input. In this case it was an IRS payment entry system that >>> would skip or duplicate button presses. >>> >>> Our setup is pretty generic: >>> Polycom SoundPoint IP 320 SIP -> dedicated LAN -> Freeswitch (on CentOS, >>> on XenServer) -> Comcast static IP -> Flowroute.com -> some IRS 800 number. >>> >>> Is this just the way it is, or is there something I can do to make it >>> work better? >>> >>> -- >>> Steve >>> >> > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/750e6ac2/attachment.html From jaykris at gmail.com Wed May 22 02:34:09 2013 From: jaykris at gmail.com (JP) Date: Tue, 21 May 2013 15:34:09 -0700 Subject: [Freeswitch-users] Separating Signaling and Media Among 2 FreeSWITCHs Message-ID: Is there any way to separate the signaling and media for a call across 2 FreeSWITCH servers? Suppose that I would ideally like to transfer a call from one FS server to another, but the user agent doesn't support any kind of SIP methods to accomplish this (ex. REFER, INVITE with "Replaces" header, etc.). Is there a way to re-route the media to another FS server while keeping the signaling anchored on the original FS? Thanks JP -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/b5f2db6a/attachment.html From msc at freeswitch.org Wed May 22 02:36:55 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 15:36:55 -0700 Subject: [Freeswitch-users] Codec Negotiation Error Lync/UCMA -> external provider In-Reply-To: <1369152691371-7590911.post@n2.nabble.com> References: <1369152691371-7590911.post@n2.nabble.com> Message-ID: Looks like a device is misbehaving and violating RFC3711. Fortunately you can explicitly allow this behavior: http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-crypto-in-avp Let us know if that works. -MC On Tue, May 21, 2013 at 9:11 AM, JohnGalt1717 wrote: > Hi there, I'm a newbee, but I've been reading everything I can and I can't > find anything that assists with this. > > I have a UCMA app that is internal that communicates on the internal sip > port (set to 5042) and it bridges with dialplan to the external sip trunk > (trying 2, one of which is siproutes). > > If I enable proxy, the call rings out and I get dead air on the other side, > but it won't ever properly establish the call in the UCMA app because of > some issue with codecs from what I can tell. > > Obviously I don't want to use proxy mode, what I really want to do is use > it > straight up and have FS do the transcoding etc. > > The problem is that once I do this I get a Codec Negotiation error and I > can't figure out why. Anyone have any ideas as to why it's freaking out > like > this? (log below) Thanks! > > 2013-05-21 12:09:41.570884 [DEBUG] sofia_glue.c:1219 Local SDP: > v=0 > o=FreeSWITCH 1369129003 1369129004 IN IP4 10.8.0.158 > s=FreeSWITCH > c=IN IP4 10.8.0.158 > t=0 0 > m=audio 23578 RTP/AVP 9 0 8 101 13 > a=rtpmap:9 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:114 > (sofia/external/+16032897972) State Change CS_INIT -> CS_ROUTING > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/+16032897972 [BREAK] > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/+16032897972) State INIT going to sleep > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+16032897972) Running State Change CS_ROUTING > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+16032897972) State ROUTING > 2013-05-21 12:09:41.570884 [DEBUG] mod_sofia.c:137 > sofia/external/+16032897972 SOFIA ROUTING > 2013-05-21 12:09:41.570884 [DEBUG] switch_ivr_originate.c:67 > (sofia/external/+16032897972) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/+16032897972 [BREAK] > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/+16032897972) State ROUTING going to sleep > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/+16032897972) Running State Change CS_CONSUME_MEDIA > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/+16032897972) State CONSUME_MEDIA > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/+16032897972) State CONSUME_MEDIA going to sleep > 2013-05-21 12:09:41.570884 [DEBUG] switch_core_session.c:1005 Send signal > sofia/external/+16032897972 [BREAK] > 2013-05-21 12:09:41.570884 [DEBUG] sofia.c:5742 Channel > sofia/external/+16032897972 entering state [calling][0] > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal > sofia/external/+16032897972 [BREAK] > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_session.c:1005 Send signal > sofia/external/+16032897972 [BREAK] > 2013-05-21 12:09:43.891308 [INFO] sofia.c:966 sofia/external/+16032897972 > Update Callee ID to "+16032897972" > 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5742 Channel > sofia/external/+16032897972 entering state [proceeding][183] > 2013-05-21 12:09:43.891308 [DEBUG] sofia.c:5751 Remote SDP: > v=0 > o=Sonus_UAC 10608 14579 IN IP4 207.239.33.59 > s=SIP Media Capabilities > c=IN IP4 207.239.33.59 > t=0 0 > m=audio 10460 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=maxptime:20 > > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec > Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2714 Audio Codec > Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:1813 Set Codec > sofia/external/+16032897972 PCMU/8000 20 ms 160 samples 64000 bits > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_codec.c:111 > sofia/external/+16032897972 Original read codec set to PCMU:0 > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:2876 Set 2833 dtmf > send payload to 101 > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:3800 AUDIO RTP > [sofia/external/+16032897972] 10.8.0.158 port 23578 -> 207.239.33.59 port > 10460 codec: 0 m > s: 20 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:2659 Starting timer [soft] > 160 bytes per 20ms > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: START > SESSION INITIALIZATION. sID=4. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > ZID=306130383030396530393363. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Loading User's profile: > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > allowclear: OFF > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > autosecure: ON > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > disclose_bit: OFF > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > signal. role: Initiator > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > TTL: 4294967295 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: SAS > schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 B256 > 2013-05-21 12:09: > 43.891308 [DEBUG] switch_rtp.c:1231 B32 2013-05-21 12:09:43.891308 [DEBUG] > switch_rtp.c:1231 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Ciphers: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 AES3 > 2013-05-21 12:09:4 > 3.891308 [DEBUG] switch_rtp.c:1231 AES1 2013-05-21 12:09:43.891308 [DEBUG] > switch_rtp.c:1231 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: PK > schemes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 EC25 > 2013-05-21 12:09: > 43.891308 [DEBUG] switch_rtp.c:1231 DH3k 2013-05-21 12:09:43.891308 [DEBUG] > switch_rtp.c:1231 DH2k 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 > Mult 201 > 3-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > ATL: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 HS32 2013-05-21 > 12:09: > 43.891308 [DEBUG] switch_rtp.c:1231 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Hashes: 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 S256 > 2013-05-21 > 12:09:4 > 3.891308 [DEBUG] switch_rtp.c:1231 > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Session initialization - DONE. sID=4. > > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > ATTACH > NEW STREAM to sID=4: > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: > Stream ID=0 UNKNOWN switching ---> . > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Empty slot was found - initializing new stream with ID=4. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > Preparing ZRTP Hello according to the Session profile. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp main]: > ATTACH > NEW STREAM - DONE. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp engine]: START > STREAM ID=4 mode=CLEAR state=ACTIVE. > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp]: > Stream ID=4 CLEAR switching ---> . > 2013-05-21 12:09:43.891308 [DEBUG] switch_rtp.c:1231 [ zrtp utils]: > Send > ssrc=1428962591 seq=54611 size=144. Stream 4:CLEAR:START > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4145 Set 2833 dtmf > send payload to 101 > 2013-05-21 12:09:43.891308 [DEBUG] switch_core_media.c:4151 Set 2833 dtmf > receive payload to 101 > 2013-05-21 12:09:43.891308 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/+16032897972! > 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3273 Send signal > sofia/internal/+16032897972 at 173.255.174.20 [BREAK] > 2013-05-21 12:09:43.891308 [DEBUG] switch_channel.c:3277 > (sofia/external/+16032897972) Callstate Change DOWN -> EARLY > 2013-05-21 12:09:43.911297 [DEBUG] switch_ivr_originate.c:411 Setting codec > string on sofia/internal/+16032897972 at 173.255.174.20 to PCMU at 8000h@20i > 2013-05-21 12:09:43.911297 [INFO] switch_ivr_originate.c:3416 Sending early > media > 2013-05-21 12:09:43.911297 [ERR] switch_core_media.c:2549 a=crypto in > RTP/AVP, refer to rfc3711 > 2013-05-21 12:09:43.911297 [ERR] mod_sofia.c:2081 CODEC NEGOTIATION ERROR. > SDP: > v=0 > o=- 1 0 IN IP4 10.8.0.158 > s=session > c=IN IP4 10.8.0.158 > b=CT:10000 > t=0 0 > m=audio 2908 RTP/AVP 9 112 111 0 8 116 13 118 97 101 > c=IN IP4 10.8.0.158 > a=rtpmap:9 G722/8000 > a=rtpmap:112 G7221/16000 > a=fmtp:112 bitrate=24000 > a=rtpmap:111 SIREN/16000 > a=fmtp:111 bitrate=16000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:116 AAL2-G726-32/8000 > a=rtpmap:13 CN/8000 > a=rtpmap:118 CN/16000 > a=rtpmap:97 RED/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16,36 > a=rtcp:2909 > a=label:main-audio > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:BqtEfRv/sMasd3GS0I4FJRX5hh7BKVt+r8/vKbOk > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Codec-Negotiation-Error-Lync-UCMA-external-provider-tp7590911.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://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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/c32f4d2b/attachment-0001.html From krice at freeswitch.org Wed May 22 02:52:41 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 21 May 2013 17:52:41 -0500 Subject: [Freeswitch-users] Separating Signaling and Media Among 2 FreeSWITCHs In-Reply-To: Message-ID: You can bypass media mode or something like that But the second server will still be in the sip path... How else are you going to tell FS heres a call On 5/21/13 5:34 PM, "JP" wrote: > Is there any way to separate the signaling and media for a call across 2 > FreeSWITCH servers? > > Suppose that I would ideally like to transfer a call from one FS server to > another, but the user agent doesn't support any kind of SIP methods to > accomplish this (ex. REFER, INVITE with "Replaces" header, etc.).? Is there a > way to re-route the media to another FS server while keeping the signaling > anchored on the original FS? > > Thanks > > JP? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/3259727f/attachment.html From jaykris at gmail.com Wed May 22 03:22:59 2013 From: jaykris at gmail.com (JP) Date: Tue, 21 May 2013 16:22:59 -0700 Subject: [Freeswitch-users] Separating Signaling and Media Among 2 FreeSWITCHs In-Reply-To: References: Message-ID: Thanks!. Let me try that. -JP On Tue, May 21, 2013 at 3:52 PM, Ken Rice wrote: > You can bypass media mode or something like that > But the second server will still be in the sip path... How else are you > going to tell FS heres a call > > > > On 5/21/13 5:34 PM, "JP" wrote: > > Is there any way to separate the signaling and media for a call across 2 > FreeSWITCH servers? > > Suppose that I would ideally like to transfer a call from one FS server to > another, but the user agent doesn't support any kind of SIP methods to > accomplish this (ex. REFER, INVITE with "Replaces" header, etc.). Is there > a way to re-route the media to another FS server while keeping the > signaling anchored on the original FS? > > Thanks > > JP > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/45004212/attachment.html From schoch+freeswitch.org at xwin32.com Wed May 22 03:41:22 2013 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 21 May 2013 16:41:22 -0700 Subject: [Freeswitch-users] Notification of an event in the dialplan In-Reply-To: References: Message-ID: On Tue, May 21, 2013 at 2:26 PM, Michael Collins wrote: > > > It's a little off the subject, but I use luarun in my dialplan to do a CNAM lookup in the background while an IVR is waiting for the caller to make a choice. Luarun runs a LUA script in the background. -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/dbcf5f94/attachment.html From jaganthoutam at gmail.com Wed May 22 03:43:43 2013 From: jaganthoutam at gmail.com (Jagadish Thoutam) Date: Tue, 21 May 2013 19:43:43 -0400 Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error Message-ID: Hi All, FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Error : while error its not accepting calls o=Sonus_UAC 902944909 319279515 IN IP4 YY.YY.YY.142 s=SIP Media Capabilities c=IN IP4 YY.YY.YY.131 t=0 0 m=audio 28390 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=maxptime:20 2013-05-21 11:09:13.488398 [NOTICE] sofia.c:5799 Pre-Answer sofia/external/17863565651! 2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3273 Send signal sofia/external/XXXXXXXXXX at 192.168.10.123 [BREAK] 2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3277 (sofia/external/17863565651) Callstate Change DOWN -> EARLY 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:1720 Set Codec sofia/external/17863565651 PROXY/8000 20 ms 160 samples 0 bits 2013-05-21 11:09:13.488398 [DEBUG] switch_core_codec.c:111 sofia/external/17863565651 Original read codec set to PROXY:0 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:3746 PROXY AUDIO RTP [sofia/external/17863565651] YY.YY.YY.131:28390->YY.YY.YY.131:28390 codec: 0 ms: 20 2013-05-21 11:09:13.488398 [DEBUG] switch_rtp.c:2667 Not using a timer 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4025 Set 2833 dtmf send payload to 101 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4031 Set 2833 dtmf receive payload to 101 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4064 Set comfort noise payload to 13 2013-05-21 11:09:13.488398 [DEBUG] switch_core_session.c:922 Send signal sofia/external/XXXXXXXXXX at 192.168.10.123 [BREAK] 2013-05-21 11:09:13.508407 [DEBUG] switch_ivr_originate.c:414 Codec string PROXY at 8000h@20i not supported on sofia/external/XXXXXXXXXX at 192.168.10.123, skipping inheritance 2013-05-21 11:09:13.508407 [INFO] switch_ivr_originate.c:3417 Sending early media 2013-05-21 11:09:13.508407 [CRIT] switch_core_media.c:3367 No audio RTP ports available! 2013-05-21 11:09:13.508407 [ERR] switch_core_media.c:5757 sofia/external/ XXXXXXXXXX at 192.168.10.123 I/O Error -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/09863b61/attachment.html From sdevoy at bizfocused.com Wed May 22 05:12:55 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 21 May 2013 21:12:55 -0400 Subject: [Freeswitch-users] One Way Audio In-Reply-To: References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> Message-ID: <027601ce5689$7de60d20$79b22760$@bizfocused.com> Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/1f22edf0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/1f22edf0/attachment-0001.gif From philippe at ppmt.org Wed May 22 05:46:41 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 21 May 2013 21:46:41 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: <519C2381.7070603@ppmt.org> your inbound is properly configure (well it is registered at least) so it should be able to send any inbound call to your FS server. As MC explained it much better than I did I am not going to try to confuse you but the magic is in the dialplan! I have been using FS for some time now but only using it like you to make and receive calls so I am not exactly an expert. One advice I can give you is to read the Wiki (http://wiki.freeswitch.org/wiki/Main_Page) It contains a lot of example and you will most likely find inspiration there and of course the Freeswitch book is a must to get even more detailed information. You can buy it form the FS website(http://www.freeswitch.org/) On 13-05-21 01:47 PM, Mike Hendrie wrote: > > As you may have guessed, I'm new to the application. Can you say how I > should configure the inbound? > > Mike > > On May 21, 2013 12:43 PM, "Philippe Le Toquin" > wrote: > > For the inbound I didn't comment because I can't understand how > your dialplan would link your gateway to your extension > > > > > On 21 May 2013 00:50, Mike Hendrie > wrote: > > I will take a closer look at the gateways. However, your > email looks like it was sent before you completed it. > > What about the inbound DID? Any ideas? > > > On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin > > wrote: > > I am no expert so I could be wrong but: > > First your outbound is no registered (NOREG) so that is > most likely going to prevent outgoing call > > Also you refer to default_gateway in your dialplan....what > value is it set to? Check vars.xmls > > I normally use directly the name of my gateway instead of > using variable (might not be the best way though ) > > > > But you have the > On 13-05-19 11:39 PM, Mike Hendrie wrote: >> Thank you for your response. Per your request here is >> the result of the sofia status command: >> >> freeswitch at internal> sofia status >> Name Type Data >> State >> ================================================================================================= >> 10.1.1.5 alias internal >> ALIASED >> internal profile >> sip:mod_sofia at 10.1.1.5:5060 >> RUNNING (0) >> external profile >> sip:mod_sofia at 10.1.1.5:5080 >> RUNNING (0) >> external::example.com >> gateway sip:joeuser at example.com >> NOREG >> external::BatCave-inbound gateway >> sip:J0k3R!@BacCave.hole.Xom >> REGED >> external::BatCat-outbound gateway >> sip:J0k3R!@BacCave.hole.Xom >> NOREG >> GothamCity.xom alias internal >> ALIASED >> internal-ipv6 profile >> sip:mod_sofia@[::1]:5060 RUNNING (0) >> ================================================================================================= >> 3 profiles 2 aliases >> >> >> ========================================== >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> >> >> >> >> ========================================== >> >> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >> >> >> >> >> >> > expression="^(\d{7})$"> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >> >> >> >> >> >> > expression="^(\d{11})$"> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> > expression="international"/> >> > expression="^(011\d+)$"> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="sofia/gateway/${default_gateway}/$1"/> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >> >> >> >> >> >> >> >> >> >> >> >> >> > value="outbound1.BatCave.net >> "/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ========================================== >> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >> >> >> >> >> >> >> >> > value="domestic,international,local"/> >> >> >> > value="Extension 1000"/> >> > value="1000"/> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> ========================================== >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/4da6ab9c/attachment-0001.html From anthony.minessale at gmail.com Wed May 22 07:06:18 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 21 May 2013 22:06:18 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: You should consider centos6 or debian stable. Make sure the host kernel is very new to get maximum results. On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky wrote: > Hello, > > Do you have any recommendations regarding how to set up correctly (for > production) CentOS5 openvz and FS 1.2.stable? Is there any trick to tuneup > the system to be rock solid? > Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS > 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that > something is not well configured around timers, however mod_posix_timer did > not help anything (running FS with -rp). We use our own bare metal and can > reproduce those delatas eirher when only one VE is on the HW. > Maybe time to check out centos6 with openvz? > > Any idea, recommendation, experience can be very helpful. > > Regards, > Jalsot > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/5c099863/attachment.html From bdfoster at davri.com Wed May 22 07:12:48 2013 From: bdfoster at davri.com (Brian Foster) Date: Tue, 21 May 2013 23:12:48 -0400 Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error In-Reply-To: References: Message-ID: Please use http://pastebin.freeswitch.org to copy /usr/local/freeswitch/conf/switch.conf.xml (if you're on a Linux machine). - BDF Thank you, Brian Foster Project Manager/Owner's Representative Davri Investments, Incorporated P: +1-317-787-2686 M: +1-317-600-9753 Indianapolis, Indiana On Tue, May 21, 2013 at 7:43 PM, Jagadish Thoutam wrote: > Hi All, > > FreeSWITCH Version 1.5.1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d > 2013-04-23 19:49:07Z) > > Error : while error its not accepting calls > > > > o=Sonus_UAC 902944909 319279515 IN IP4 YY.YY.YY.142 > s=SIP Media Capabilities > c=IN IP4 YY.YY.YY.131 > t=0 0 > m=audio 28390 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=maxptime:20 > > 2013-05-21 11:09:13.488398 [NOTICE] sofia.c:5799 Pre-Answer sofia/external/ > 17863565651! > 2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3273 Send signal > sofia/external/XXXXXXXXXX at 192.168.10.123 [BREAK] > 2013-05-21 11:09:13.488398 [DEBUG] switch_channel.c:3277 (sofia/external/ > 17863565651) Callstate Change DOWN -> EARLY > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:1720 Set Codec > sofia/external/17863565651 PROXY/8000 20 ms 160 samples 0 bits > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_codec.c:111 sofia/external/ > 17863565651 Original read codec set to PROXY:0 > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:3746 PROXY AUDIO > RTP [sofia/external/17863565651] YY.YY.YY.131:28390->YY.YY.YY.131:28390 > codec: 0 ms: 20 > 2013-05-21 11:09:13.488398 [DEBUG] switch_rtp.c:2667 Not using a timer > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4025 Set 2833 dtmf > send payload to 101 > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4031 Set 2833 dtmf > receive payload to 101 > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_media.c:4064 Set comfort > noise payload to 13 > 2013-05-21 11:09:13.488398 [DEBUG] switch_core_session.c:922 Send signal > sofia/external/XXXXXXXXXX at 192.168.10.123 [BREAK] > 2013-05-21 11:09:13.508407 [DEBUG] switch_ivr_originate.c:414 Codec string > PROXY at 8000h@20i not supported on sofia/external/XXXXXXXXXX at 192.168.10.123, > skipping inheritance > 2013-05-21 11:09:13.508407 [INFO] switch_ivr_originate.c:3417 Sending > early media > 2013-05-21 11:09:13.508407 [CRIT] switch_core_media.c:3367 No audio RTP > ports available! > 2013-05-21 11:09:13.508407 [ERR] switch_core_media.c:5757 sofia/external/ > XXXXXXXXXX at 192.168.10.123 I/O Error > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/f20e266d/attachment.html From thomas.lee at octon.net Wed May 22 07:35:14 2013 From: thomas.lee at octon.net (Thomas Lee) Date: Tue, 21 May 2013 20:35:14 -0700 (PDT) Subject: [Freeswitch-users] Node.JS ESL libraries In-Reply-To: <518D2B52.1090304@digitalmail.com> References: <518D2B52.1090304@digitalmail.com> Message-ID: <1369193714184-7590938.post@n2.nabble.com> Hi, Please take a look. I'm using node.js and esl (https://github.com/shimaore/esl) For example, -------------------------------------------------- // // esl-test1.js // var esl = require('esl'); var util = require('util'); var audio_file = []; // // play and get digits // // audio_file[0]='1 1 3 5000 # sounds/mywav/0.wav'; audio_file[1]='1 1 3 5000 # sounds/mywav/1.wav'; audio_file[2]='1 1 3 5000 # sounds/mywav/2.wav'; audio_file[3]='1 1 3 5000 # sounds/mywav/3.wav'; audio_file[4]='1 1 3 5000 # sounds/mywav/4.wav'; audio_file[5]='1 1 3 5000 # sounds/mywav/5.wav'; audio_file[6]='1 1 3 5000 # sounds/mywav/6.wav'; audio_file[7]='1 1 3 5000 # sounds/mywav/7.wav'; audio_file[8]='1 1 3 5000 # sounds/mywav/8.wav'; audio_file[9]='1 1 3 5000 # sounds/mywav/9.wav'; audio_file[10]='1 1 3 5000 # sounds/mywav/10.wav'; audio_file[11]='1 1 3 5000 # sounds/mywav/11.wav'; audio_file[12]='1 1 3 5000 # sounds/mywav/12.wav'; audio_file[13]='1 1 3 5000 # sounds/mywav/13.wav'; var error3 = '1 1 3 5000 # sounds/mywav/error3.wav'; var welcome = '1 1 3 5000 # sounds/mywav/hello.wav'; var server = esl.createCallServer(); //var server = esl.createClient(); server.on('CONNECT', function(req, res) { var uri, channel_data, unique_id; channel_data = req.body; unique_id = channel_data['Unique-ID']; req.execute('answer'); //req.execute('playback', 'sounds/mywav/hello.wav'); //req.execute('play_and_get_digits', '2 5 3 7000 # sounds/mywav/hello.wav sounds/mywav/error3.wav myFoo \d+'); req.execute('play_and_get_digits', welcome); req.on('DTMF', function(req) { //util.log('DTMF:'+util.inspect(req, null, null)); var digit; var channel_data; channel_data = req.body; unique_id = channel_data['Unique-ID']; util.log('DTMF: unique_id='+unique_id); digit = channel_data['DTMF-Digit']; console.log('DTMF Received=' + digit); util.log('DTMF Received'); if(digit==='#' || digit==='*'){ req.execute('play_and_get_digits', error3); } else { var n = parseInt(digit); req.execute('play_and_get_digits', audio_file[n]); } return; }); req.on('CHANNEL_ANSWER', function(req) { //util.log('CHANNEL_ANSWER:'+util.inspect(req, null, null)); var channel_data; channel_data = req.body; unique_id = channel_data['Unique-ID']; util.log('CHANNEL_ANSWER: unique_id='+unique_id); return util.log('Call was answered'); }); req.on('CHANNEL_HANGUP', function(req) { //util.log('CHANNEL_HANGUP:'+util.inspect(req, null, null)); var channel_data; channel_data = req.body; unique_id = channel_data['Unique-ID']; util.log('CHANNEL_HANGUP: unique_id='+unique_id); console.log('CHANNEL_HANGUP'); return util.log('CHANNEL_HANGUP'); }); req.on('CHANNEL_HANGUP_COMPLETE', function(req) { ///util.log('CHANNEL_HANGUP_COMPLETE:'+util.inspect(req, null, null)); var channel_data; channel_data = req.body; unique_id = channel_data['Unique-ID']; util.log('CHANNEL_HANGUP_COMPLETE: unique_id='+unique_id); console.log('CHANNEL_HANGUP_COMPLETE'); return util.log('CHANNEL_HANGUP_COMPLETE'); }); req.on('DISCONNECT', function(req) { //util.log('DISCONNECT:'+util.inspect(req, null, null)); var channel_data; channel_data = req.body; unique_id = channel_data['Unique-ID']; util.log('DISCONNECT: unique_id='+unique_id); console.log('DISCONNECT'); return util.log('DISCONNECT'); }) //util.log('CONNECT: req '+util.inspect(req, null, null)); util.log('CONNECT: unique_id='+unique_id); return util.log('CONNECT received'); }); server.listen(9173); ----------------------------------------------- Thanks Regards, Thomas Lee -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Node-JS-ESL-libraries-tp7590562p7590938.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ashish at nms.co.in Wed May 22 09:04:57 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 22 May 2013 10:34:57 +0530 Subject: [Freeswitch-users] Switch on/off call recording Message-ID: Hi, I have setup a little call center application using mod_callcenter. Now what I want is to record all the calls based on the click of an ON?OFF button on the web app. Is there any variable or something like that which I can set like True or False to switch the call recording on or off? Also I want to know if the recording is possible per agent or not i.e. the call session to be recorded for a particular agent. Please throw some light on this. Thanks in advance. Regards -Ashish -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/53807712/attachment.html From mike at hendrienet.com Wed May 22 06:27:52 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Tue, 21 May 2013 21:27:52 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Thank you for your assistance. I made the suggested modification below, however, when calling the number it goes directly to voicemail. /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml On Tue, May 21, 2013 at 4:37 PM, Michael Collins wrote: > Hi Mike, > > No worries - this can be frustrating at first. You'll get the hang of it > shortly. > > Okay, technically speaking there is no explicit link between a gateway and > an extension. Rather, the inbound call is handled by the dialplan. If > you're using the vanilla example configs (which you should be to start out) > then your inbound call is handled in the public dialplan context. This > means that you need to have a file in conf/dialplan/public/ to handle your > inbound call. It looks to me like your GothamCity.xom.xml file would work, > but you'll need to make at least two changes: > > #1 - move the file from the conf/dialplan/default/ directory into > conf/dialplan/public/ > #2 - add the regular expression for your DID number. For example, if your > DID number was 800-555-1212 you would do something like this: > > > > > > > > > Hope this helps. > -MC > > > On Tue, May 21, 2013 at 1:23 PM, Mike Hendrie wrote: > >> I think I am missing the basic understanding of how the DID connects to >> the gateway then to the extension. Which file does what. >> >> thanks!! >> >> >> On Tue, May 21, 2013 at 3:20 PM, Mike Hendrie wrote: >> >>> I am looking in the vars.xml file and am not seeing any reference to a >>> gateway: >>> >>> ====== >>> vars.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> >>> >>> >>> >> data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> >>> >>> >> data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> >>> >>> >> data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: >>> >>>> I am no expert so I could be wrong but: >>>> >>>> First your outbound is no registered (NOREG) so that is most likely >>>> going to prevent outgoing call >>>> >>>> Also you refer to default_gateway in your dialplan....what value is it >>>> set to? Check vars.xmls >>>> >>>> I normally use directly the name of my gateway instead of using >>>> variable (might not be the best way though ) >>>> >>>> >>>> >>>> But you have the >>>> On 13-05-19 11:39 PM, Mike Hendrie wrote: >>>> >>>> Thank you for your response. Per your request here is the result of >>>> the sofia status command: >>>> >>>> freeswitch at internal> sofia status >>>> Name Type >>>> Data State >>>> >>>> ================================================================================================= >>>> 10.1.1.5 alias >>>> internal ALIASED >>>> internal profile >>>> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >>>> external profile >>>> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >>>> external::example.com gateway >>>> sip:joeuser at example.com NOREG >>>> external::BatCave-inbound gateway sip:J0k3R!@BacCave.hole.Xom REGED >>>> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >>>> NOREG >>>> GothamCity.xom alias >>>> internal ALIASED >>>> internal-ipv6 profile sip:mod_sofia@[::1]:5060 >>>> RUNNING (0) >>>> >>>> ================================================================================================= >>>> 3 profiles 2 aliases >>>> >>>> >>>> ========================================== >>>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ========================================== >>>> >>>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>> >>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>> >>> data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>> >>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>> >>> data="sofia/gateway/${default_gateway}/$1"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>> >>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>> >>> data="sofia/gateway/${default_gateway}/$1"/> >>>> >>>> >>>> >>>> >>>> >>>> ========================================== >>>> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ========================================== >>>> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${outbound_caller_name}"/> >>>> >>> value="$${outbound_caller_id}"/> >>>> >>>> >>>> >>>> >>>> >>>> ========================================== >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130521/e5014131/attachment-0001.html From mike at hendrienet.com Wed May 22 08:35:15 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Tue, 21 May 2013 23:35:15 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Correction: I had a second dialplan in the public folder that was causing confusion. Below is the dialplan I am using. If I change the extension in the dialplan from 1000 to 1001 I get the appropriate voice mail extension, however, the phones never ring. I have the fs configured as a multi-tenant solution. Could the dialplan be using the default extensions (1000 and 1001) under /conf/directory/default and not reference the /conf/directory/GothamCity.xom domain? That would explain why I get to the voicemail for the correct extension when the phone never rings. Thanks ===== /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml ===== /conf/directory/GothamCity.xom.xml ===== On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: > Thank you for your assistance. I made the suggested modification below, > however, when calling the number it goes directly to voicemail. > > /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml > > > > > > > > > > > > > > On Tue, May 21, 2013 at 4:37 PM, Michael Collins wrote: > >> Hi Mike, >> >> No worries - this can be frustrating at first. You'll get the hang of it >> shortly. >> >> Okay, technically speaking there is no explicit link between a gateway >> and an extension. Rather, the inbound call is handled by the dialplan. If >> you're using the vanilla example configs (which you should be to start out) >> then your inbound call is handled in the public dialplan context. This >> means that you need to have a file in conf/dialplan/public/ to handle your >> inbound call. It looks to me like your GothamCity.xom.xml file would work, >> but you'll need to make at least two changes: >> >> #1 - move the file from the conf/dialplan/default/ directory into >> conf/dialplan/public/ >> #2 - add the regular expression for your DID number. For example, if your >> DID number was 800-555-1212 you would do something like this: >> >> >> >> >> >> >> >> >> Hope this helps. >> -MC >> >> >> On Tue, May 21, 2013 at 1:23 PM, Mike Hendrie wrote: >> >>> I think I am missing the basic understanding of how the DID connects to >>> the gateway then to the extension. Which file does what. >>> >>> thanks!! >>> >>> >>> On Tue, May 21, 2013 at 3:20 PM, Mike Hendrie wrote: >>> >>>> I am looking in the vars.xml file and am not seeing any reference to a >>>> gateway: >>>> >>>> ====== >>>> vars.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="hk-ring=%(400,200,440,480);%(400,3000,440,480)"/> >>>> >>>> >>>> >>> data="in-ring=%(400,200,425,375);%(400,2000,425,375)"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> data="uk-ring=%(400,200,400,450);%(400,2000,400,450)"/> >>>> >>>> >>> data="bong-ring=v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1400,0,350,440)"/> >>>> >>>> >>> data="sit=%(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7)"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Mon, May 20, 2013 at 4:49 PM, Philippe Le Toquin wrote: >>>> >>>>> I am no expert so I could be wrong but: >>>>> >>>>> First your outbound is no registered (NOREG) so that is most likely >>>>> going to prevent outgoing call >>>>> >>>>> Also you refer to default_gateway in your dialplan....what value is it >>>>> set to? Check vars.xmls >>>>> >>>>> I normally use directly the name of my gateway instead of using >>>>> variable (might not be the best way though ) >>>>> >>>>> >>>>> >>>>> But you have the >>>>> On 13-05-19 11:39 PM, Mike Hendrie wrote: >>>>> >>>>> Thank you for your response. Per your request here is the result of >>>>> the sofia status command: >>>>> >>>>> freeswitch at internal> sofia status >>>>> Name Type >>>>> Data State >>>>> >>>>> ================================================================================================= >>>>> 10.1.1.5 alias >>>>> internal ALIASED >>>>> internal profile >>>>> sip:mod_sofia at 10.1.1.5:5060 RUNNING (0) >>>>> external profile >>>>> sip:mod_sofia at 10.1.1.5:5080 RUNNING (0) >>>>> external::example.com gateway >>>>> sip:joeuser at example.com NOREG >>>>> external::BatCave-inbound gateway >>>>> sip:J0k3R!@BacCave.hole.Xom REGED >>>>> external::BatCat-outbound gateway sip:J0k3R!@BacCave.hole.Xom >>>>> NOREG >>>>> GothamCity.xom alias >>>>> internal ALIASED >>>>> internal-ipv6 profile >>>>> sip:mod_sofia@[::1]:5060 RUNNING (0) >>>>> >>>>> ================================================================================================= >>>>> 3 profiles 2 aliases >>>>> >>>>> >>>>> ========================================== >>>>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ========================================== >>>>> >>>>> /usr/local/freeswitch/conf/dialplan/default/GothamCity.xom.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>>> >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>>> >>>> data="sofia/gateway/${default_gateway}/1${default_areacode}$1"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>>> >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>>> >>>> data="sofia/gateway/${default_gateway}/$1"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>>>> >>>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>>>> >>>> data="sofia/gateway/${default_gateway}/$1"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ========================================== >>>>> /usr/local/freeswitch/conf/sip_profiles/external/vitelity.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ========================================== >>>>> /usr/local/freeswitch/conf/directory/GothamCity.xom/1000.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="domestic,international,local"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="$${outbound_caller_name}"/> >>>>> >>>> value="$${outbound_caller_id}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ========================================== >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>>> >>>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130521/42f4163f/attachment-0001.html From msc at freeswitch.org Wed May 22 10:20:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 23:20:02 -0700 Subject: [Freeswitch-users] Switch on/off call recording In-Reply-To: References: Message-ID: Are you currently recording calls now? If so, how are you doing that? -MC On Tue, May 21, 2013 at 10:04 PM, Ashish gautam wrote: > Hi, > > I have setup a little call center application using mod_callcenter. Now > what I want is to record all the calls based on the click of an ON?OFF > button on the web app. Is there any variable or something like that which I > can set like True or False to switch the call recording on or off? > > Also I want to know if the recording is possible per agent or not i.e. the > call session to be recorded for a particular agent. > > Please throw some light on this. > > Thanks in advance. > > Regards > -Ashish > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/e5f1c308/attachment.html From ashish at nms.co.in Wed May 22 10:29:21 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 22 May 2013 11:59:21 +0530 Subject: [Freeswitch-users] Switch on/off call recording In-Reply-To: References: Message-ID: I am doing it through dialplan app record_session. And it records for all the calls. Now I want it for selected calls only. On Wed, May 22, 2013 at 11:50 AM, Michael Collins wrote: > Are you currently recording calls now? If so, how are you doing that? > -MC > > > On Tue, May 21, 2013 at 10:04 PM, Ashish gautam wrote: > >> Hi, >> >> I have setup a little call center application using mod_callcenter. Now >> what I want is to record all the calls based on the click of an ON?OFF >> button on the web app. Is there any variable or something like that which I >> can set like True or False to switch the call recording on or off? >> >> Also I want to know if the recording is possible per agent or not i.e. >> the call session to be recorded for a particular agent. >> >> Please throw some light on this. >> >> Thanks in advance. >> >> Regards >> -Ashish >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/ae8d4d6d/attachment.html From msc at freeswitch.org Wed May 22 10:39:39 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 23:39:39 -0700 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Post a FreeSWITCH debug log of the incoming call. Use pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax highlighting. Paste the URL in this email thread and we'll take a look. -MC On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: > Correction: > I had a second dialplan in the public folder that was causing confusion. > Below is the dialplan I am using. > If I change the extension in the dialplan from 1000 to 1001 I get > the appropriate voice mail extension, however, the phones never ring. > > I have the fs configured as a multi-tenant solution. > > Could the dialplan be using the default extensions (1000 and 1001) under > /conf/directory/default and not reference the > /conf/directory/GothamCity.xom domain? That would explain why I get to the > voicemail for the correct extension when the phone never rings. > > Thanks > > ===== > /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml > > > > > > > > > > > > > ===== > /conf/directory/GothamCity.xom.xml > > > > > value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ > ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > ===== > > > > > On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: > >> Thank you for your assistance. I made the suggested modification below, >> however, when calling the number it goes directly to voicemail. >> >> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >> >> >> >> >> >> >> >> >> >> >> >> -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/0a2a069c/attachment-0001.html From ehermouet at bluetel.fr Wed May 22 10:44:16 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Wed, 22 May 2013 08:44:16 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> Message-ID: <2a3eb9d6-3c1a-4825-9988-a8a357ffa433@email.android.com> Brian did you find something on log file ? Maybe i miss config option. Tks Brian Foster a ?crit?: >No, you will not need to use that. >On May 21, 2013 8:00 AM, wrote: > >> I must use ? if yes where >> and how ? >> >> tks advance for your help >> >> >> Le 2013-05-20 23:11, Brian Foster a ?crit : >> > 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 >mode >> > may not work on a transcoded call. >> > >> > You shouldnt be transcoding if you can help it. Now, Im not sure if >> > that is an empty threat but you should enable late codec >negotiation. >> > Information can be found here: >> > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] >> > >> > -BDF >> > On May 17, 2013 3:48 AM, wrote: >> > >> >> im so stupid :) >> >> tks >> >> >> >> http://pastebin.freeswitch.org/20933 [1] >> >> >> >> called num is 022206... and when i try to use dtmf touch 5 its not >> >> works. >> >> >> >> tks >> >> >> >> Le 2013-05-17 09:25, Ken Rice a ?crit : >> >> > it tells you the password in the popup... this is an anti spam >> >> thing >> >> > >> >> > KenSent from my iPad >> >> > >> >> > On May 17, 2013, at 0:46, Hermouet Erwan > >> [2] [14]> >> >> > wrote: >> >> > >> >> >> On login i try my email...but don t work...i loose here >> >> >> >> >> >> Michael Collins a ?crit : >> >> >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >> >>> wrote: >> >> >>> >> >> >>>> I have the log but i never found how works pastebin ?? do you >> >> >>>> have tutorial ? >> >> >>> >> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> >> text >> >> >>> box, select FreeSWITCH Log as the syntax highlighting and then >> >> >>> click Send. Copy the URL from the browse address bar. it will >> >> be >> >> >>> something like: >> >> >>> http://pastebin.freeswitch.org/20927 [5] [2] >> >> >>> >> >> >>> -MC >> >> >> >> >> >> Hermouet Erwan >> >> >> Responsable technique >> >> >> Bluetel >> >> > >> >> >> >> >> > >> >> > >> >> >> > >> > >_________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org [6] [4] >> >> >> http://www.freeswitchsolutions.com [7] [5] >> >> >> >> >> >> >> >> >> [8] [6] >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org [9] [7] >> >> >> http://wiki.freeswitch.org [10] [8] >> >> >> http://www.cluecon.com [11] [9] >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> [13] [11] >> >> >> >> >> > >> >> > >> >> >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [14] >> >> >> [12] >> >> >> http://www.freeswitch.org [15] [13] >> >> > >> >> > >> >> > Links: >> >> > ------ >> >> > [1] mailto:ehermouet at bluetel.fr [16] >> >> > [2] http://pastebin.freeswitch.org/20927 [17] >> >> > [3] mailto:msc at freeswitch.org [18] >> >> > [4] mailto:consulting at freeswitch.org [19] >> >> > [5] http://www.freeswitchsolutions.com [20] >> >> > [6] [21] >> >> > [7] http://www.freeswitch.org [22] >> >> > [8] http://wiki.freeswitch.org [23] >> >> > [9] http://www.cluecon.com [24] >> >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] >> >> > [11] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] >> >> > [12] >http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [27] >> >> > [13] http://www.freeswitch.org [28] >> >> > [14] mailto:ehermouet at bluetel.fr [29] >> >> >> >> >> > >> > >_________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [30] >> >> http://www.freeswitchsolutions.com [31] >> >> >> >> >> >> [32] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [33] >> >> http://wiki.freeswitch.org [34] >> >> http://www.cluecon.com [35] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [36] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] >> >> >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [38] >> >> http://www.freeswitch.org [39] >> > >> > >> > Links: >> > ------ >> > [1] http://pastebin.freeswitch.org/20933 >> > [2] mailto:ehermouet at bluetel.fr >> > [3] mailto:msc at freeswitch.org >> > [4] mailto:ehermouet at bluetel.fr >> > [5] http://pastebin.freeswitch.org/20927 >> > [6] mailto:consulting at freeswitch.org >> > [7] http://www.freeswitchsolutions.com >> > [8] >> > [9] http://www.freeswitch.org >> > [10] http://wiki.freeswitch.org >> > [11] http://www.cluecon.com >> > [12] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [15] http://www.freeswitch.org >> > [16] mailto:ehermouet at bluetel.fr >> > [17] http://pastebin.freeswitch.org/20927 >> > [18] mailto:msc at freeswitch.org >> > [19] mailto:consulting at freeswitch.org >> > [20] http://www.freeswitchsolutions.com >> > [21] >> > [22] http://www.freeswitch.org >> > [23] http://wiki.freeswitch.org >> > [24] http://www.cluecon.com >> > [25] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [28] http://www.freeswitch.org >> > [29] mailto:ehermouet at bluetel.fr >> > [30] mailto:consulting at freeswitch.org >> > [31] http://www.freeswitchsolutions.com >> > [32] >> > [33] http://www.freeswitch.org >> > [34] http://wiki.freeswitch.org >> > [35] http://www.cluecon.com >> > [36] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [39] http://www.freeswitch.org >> > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation >> > [41] mailto:ehermouet at bluetel.fr >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/72334a82/attachment.html From msc at freeswitch.org Wed May 22 10:44:45 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 21 May 2013 23:44:45 -0700 Subject: [Freeswitch-users] Switch on/off call recording In-Reply-To: References: Message-ID: You could have your web app set a channel variable and then only turn on recording if that channel variable was set to true: Just be sure to use continue="true" on that extension. ;) -MC On Tue, May 21, 2013 at 11:29 PM, Ashish gautam wrote: > I am doing it through dialplan app record_session. And it records for all > the calls. Now I want it for selected calls only. > > > On Wed, May 22, 2013 at 11:50 AM, Michael Collins wrote: > >> Are you currently recording calls now? If so, how are you doing that? >> -MC >> >> >> On Tue, May 21, 2013 at 10:04 PM, Ashish gautam wrote: >> >>> Hi, >>> >>> I have setup a little call center application using mod_callcenter. Now >>> what I want is to record all the calls based on the click of an ON?OFF >>> button on the web app. Is there any variable or something like that which I >>> can set like True or False to switch the call recording on or off? >>> >>> Also I want to know if the recording is possible per agent or not i.e. >>> the call session to be recorded for a particular agent. >>> >>> Please throw some light on this. >>> >>> Thanks in advance. >>> >>> Regards >>> -Ashish >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130521/bf27e266/attachment-0001.html From msc at freeswitch.org Wed May 22 11:11:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 00:11:01 -0700 Subject: [Freeswitch-users] One Way Audio In-Reply-To: <027601ce5689$7de60d20$79b22760$@bizfocused.com> References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> <027601ce5689$7de60d20$79b22760$@bizfocused.com> Message-ID: You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy wrote: > Hey Michael,**** > > ** ** > > The problem is back and seems to all the time now.**** > > ** ** > > Pcapsipdump for a single call is here > http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz **** > > ** ** > > I would like to know why calling FROM FS to a SPRINT phone results in > audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. > **** > > ** ** > > Also, any tips on getting started with wireshark to investigate myself > next time would be appreciated.**** > > ** ** > > Thanks,**** > > Sean**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Friday, May 17, 2013 7:29 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] One Way Audio**** > > ** ** > > It wouldn't be the first time that a computer decided to behave because it > knew Daddy was watching...**** > > -MC**** > > ** ** > > On Thu, May 16, 2013 at 7:20 PM, Sean Devoy wrote: > **** > > Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed.*** > * > > **** > > My wife had just called my cell and got one way audio. So I ran: > pcapsipdump -f -p -i eth0 -d /tmp -n **** > > Of course I got 2 way audio. I called the one that ALWAYS fails ?. Got 2 > way audio! Does pcapsipdump fix it? lol**** > > **** > > I will try in day time tomorrow and see if we can get a failure.**** > > **** > > Nothing in the freeswitch.log of value? I didn?t see anything, but there > is still a lot for me to learn there.**** > > **** > > Sean**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, May 16, 2013 7:35 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] One Way Audio**** > > **** > > Sean,**** > > Glad to hear you're making progress with using tcpdump and other packet > capture-ish tools. You've successfully captured the SIP call leg between > your phone and your FreeSWITCH. That's good, but it's incomplete. You > really want SIP and RTP both, and you want the call leg between FreeSWITCH > and the telco. You have a few options:**** > > Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. > Con: creates massive pcap files through which you must sift to find the > call in question.**** > > Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for > each call leg into a single file. Cons: You have to compile it yourself, > and it creates a whole lot of files, so you'll need to get used to it.**** > > My personal opinion is this: if you never, ever have to debug a SIP call > ever again then just use tcpdump. However, if you're the phone guy and > you'll be doing this again in the future then bite the bullet and learn > pcapsipdump. Believe me it's worth it.**** > > -MC**** > > **** > > On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: > **** > > **** > > Hi all,**** > > **** > > First, I am on version: FreeSWITCH Version > 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) > I hope to move to the Stable 1.2.9 this weekend.**** > > **** > > I am having very reliable one way audio when calling Sprint Cell Phone > users, though not every time. I got this reproducible on my phone, but by > the time I learned tcpdump command, it was working again. However, the > user that reported it seems to get it almost everytime.**** > > **** > > Helpful tidbits:**** > > ? I THINK it happens in either direction.**** > > ? For this person at his home, it appears to be every time (for > now)**** > > ? He reports calling (to or from) other Sprint Cell users results > in the same problem from our FS**** > > ? It appears to only be true with Sprint Cell calls! (But my > users say that?s not Sprints fault!)**** > > **** > > Scenario:**** > > I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and > it rings, he answers, I can hear him crystal clear ? he can?t hear me at > all. I had a theory that it worked after 30 seconds (on my cell), but that > does not hold true on his cell.**** > > **** > > Here is the FS logfile: > http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt** > ** > > And here is the tcpdump output: > http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip **** > > **** > > Based on the small size of the file, I suspect someone is going to say ?do > it again with this tcpdump command?. I welcome the education.**** > > **** > > Anyway, any insight will be appreciated.**** > > **** > > Sean**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/51fa6630/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/51fa6630/attachment.gif From garbytrash at gmail.com Wed May 22 11:56:06 2013 From: garbytrash at gmail.com (Zenny) Date: Wed, 22 May 2013 09:56:06 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: On 5/22/13, Anthony Minessale wrote: > You should consider centos6 or debian stable. Make sure the host kernel is > very new to get maximum results. Tony, do you mean "very new kernel" means 3.2.xx kernel? Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not possible. And that is what CentOS6 offers, too. However, I installed FS as openvz guest, it works fine for outgoing, but not DNAT works for incoming connections even after throroughly following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. Just my two cents. > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky wrote: > >> Hello, >> >> Do you have any recommendations regarding how to set up correctly (for >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> tuneup >> the system to be rock solid? >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> something is not well configured around timers, however mod_posix_timer >> did >> not help anything (running FS with -rp). We use our own bare metal and >> can >> reproduce those delatas eirher when only one VE is on the HW. >> Maybe time to check out centos6 with openvz? >> >> Any idea, recommendation, experience can be very helpful. >> >> Regards, >> Jalsot >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > From ehermouet at bluetel.fr Wed May 22 12:29:33 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Wed, 22 May 2013 10:29:33 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> Message-ID: So i'm the only man with this problem ? nobody use dtmf ? :'( Le 2013-05-21 14:03, Brian Foster a ?crit?: > No, you will not need to use that. > On May 21, 2013 8:00 AM, wrote: > >> I must use ? if yes >> where >> and how ? >> >> tks advance for your help >> >> Le 2013-05-20 23:11, Brian Foster a ?crit?: >> > 2013-05-17?09 >> [1]:39:54.214928?[WARNING]?mod_sofia.c:1363?Pass?2833?mode >> > may not work on a transcoded call. >> > >> > You shouldnt be transcoding if you can help it. Now, Im not sure >> if >> > that is an empty threat but you should enable late codec >> negotiation. >> > Information can be found here: >> > http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >> > >> > -BDF >> > On May 17, 2013 3:48 AM, wrote: >> > >> >> im so stupid :) >> >> tks >> >> >> >> http://pastebin.freeswitch.org/20933 [4] [1] >> >> >> >> called num is 022206... and when i try to use dtmf touch 5 its >> not >> >> works. >> >> >> >> tks >> >> >> >> Le 2013-05-17 09:25, Ken Rice a ?crit?: >> >> > it tells you the password in the popup... this is an anti spam >> >> thing >> >> > >> >> > KenSent from my iPad >> >> > >> >> > On May 17, 2013, at 0:46, Hermouet Erwan > [5] >> >> [2] [14]> >> >> > wrote: >> >> > >> >> >> On login i try my email...but don t work...i loose here >> >> >> >> >> >> Michael Collins a ?crit : >> >> >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >> >>> wrote: >> >> >>> >> >> >>>> I have the log but i never found how works pastebin ?? do >> you >> >> >>>> have tutorial ? >> >> >>> >> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> >> text >> >> >>> box, select FreeSWITCH Log as the syntax highlighting and >> then >> >> >>> click Send. Copy the URL from the browse address bar. it >> will >> >> be >> >> >>> something like: >> >> >>> http://pastebin.freeswitch.org/20927 [8] [5] [2] >> >> >>> >> >> >>> -MC >> >> >> >> >> >> Hermouet Erwan >> >> >> Responsable technique >> >> >> Bluetel >> >> > >> >> >> >> >> > >> >> > >> >> >> > >> > >> > > _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org [9] [6] [4] >> >> >> http://www.freeswitchsolutions.com [10] [7] [5] >> >> >> >> >> >> >> >> >> [11] [8] [6] >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org [12] [9] [7] >> >> >> http://wiki.freeswitch.org [13] [10] [8] >> >> >> http://www.cluecon.com [14] [11] [9] >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org [15] [12] [10] >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [16] >> >> [13] [11] >> >> >> >> >> > >> >> > >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [17] >> >> [14] >> >> >> [12] >> >> >> http://www.freeswitch.org [18] [15] [13] >> >> > >> >> > >> >> > Links: >> >> > ------ >> >> > [1] mailto:ehermouet at bluetel.fr [19] [16] >> >> > [2] http://pastebin.freeswitch.org/20927 [20] [17] >> >> > [3] mailto:msc at freeswitch.org [21] [18] >> >> > [4] mailto:consulting at freeswitch.org [22] [19] >> >> > [5] http://www.freeswitchsolutions.com [23] [20] >> >> > [6] [24] [21] >> >> > [7] http://www.freeswitch.org [25] [22] >> >> > [8] http://wiki.freeswitch.org [26] [23] >> >> > [9] http://www.cluecon.com [27] [24] >> >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [28] [25] >> >> > [11] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [29] [26] >> >> > [12] >> http://lists.freeswitch.org/mailman/options/freeswitch-users [30] >> >> [27] >> >> > [13] http://www.freeswitch.org [31] [28] >> >> > [14] mailto:ehermouet at bluetel.fr [32] [29] >> >> >> >> >> > >> > >> > > _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [33] [30] >> >> http://www.freeswitchsolutions.com [34] [31] >> >> >> >> >> >> [35] [32] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [36] [33] >> >> http://wiki.freeswitch.org [37] [34] >> >> http://www.cluecon.com [38] [35] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [39] [36] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> [40] [37] >> >> >> > >> > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [41] >> >> [38] >> >> http://www.freeswitch.org [42] [39] >> > >> > >> > Links: >> > ------ >> > [1] http://pastebin.freeswitch.org/20933 [43] >> > [2] mailto:ehermouet at bluetel.fr [44] >> > [3] mailto:msc at freeswitch.org [45] >> > [4] mailto:ehermouet at bluetel.fr [46] >> > [5] http://pastebin.freeswitch.org/20927 [47] >> > [6] mailto:consulting at freeswitch.org [48] >> > [7] http://www.freeswitchsolutions.com [49] >> > [8] [50] >> > [9] http://www.freeswitch.org [51] >> > [10] http://wiki.freeswitch.org [52] >> > [11] http://www.cluecon.com [53] >> > [12] mailto:FreeSWITCH-users at lists.freeswitch.org [54] >> > [13] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [55] >> > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [56] >> > [15] http://www.freeswitch.org [57] >> > [16] mailto:ehermouet at bluetel.fr [58] >> > [17] http://pastebin.freeswitch.org/20927 [59] >> > [18] mailto:msc at freeswitch.org [60] >> > [19] mailto:consulting at freeswitch.org [61] >> > [20] http://www.freeswitchsolutions.com [62] >> > [21] [63] >> > [22] http://www.freeswitch.org [64] >> > [23] http://wiki.freeswitch.org [65] >> > [24] http://www.cluecon.com [66] >> > [25] mailto:FreeSWITCH-users at lists.freeswitch.org [67] >> > [26] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [68] >> > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [69] >> > [28] http://www.freeswitch.org [70] >> > [29] mailto:ehermouet at bluetel.fr [71] >> > [30] mailto:consulting at freeswitch.org [72] >> > [31] http://www.freeswitchsolutions.com [73] >> > [32] [74] >> > [33] http://www.freeswitch.org [75] >> > [34] http://wiki.freeswitch.org [76] >> > [35] http://www.cluecon.com [77] >> > [36] mailto:FreeSWITCH-users at lists.freeswitch.org [78] >> > [37] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [79] >> > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users >> [80] >> > [39] http://www.freeswitch.org [81] >> > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation [82] >> > [41] mailto:ehermouet at bluetel.fr [83] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [84] >> http://www.freeswitchsolutions.com [85] >> >> >> [86] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [87] >> http://wiki.freeswitch.org [88] >> http://www.cluecon.com [89] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [90] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [91] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [92] >> http://www.freeswitch.org [93] > > > Links: > ------ > [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 > [2] http://wiki.freeswitch.org/wiki/Codec_negotiation > [3] mailto:ehermouet at bluetel.fr > [4] http://pastebin.freeswitch.org/20933 > [5] mailto:ehermouet at bluetel.fr > [6] mailto:msc at freeswitch.org > [7] mailto:ehermouet at bluetel.fr > [8] http://pastebin.freeswitch.org/20927 > [9] mailto:consulting at freeswitch.org > [10] http://www.freeswitchsolutions.com > [11] > [12] http://www.freeswitch.org > [13] http://wiki.freeswitch.org > [14] http://www.cluecon.com > [15] mailto:FreeSWITCH-users at lists.freeswitch.org > [16] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [17] http://lists.freeswitch.org/mailman/options/freeswitch-users > [18] http://www.freeswitch.org > [19] mailto:ehermouet at bluetel.fr > [20] http://pastebin.freeswitch.org/20927 > [21] mailto:msc at freeswitch.org > [22] mailto:consulting at freeswitch.org > [23] http://www.freeswitchsolutions.com > [24] > [25] http://www.freeswitch.org > [26] http://wiki.freeswitch.org > [27] http://www.cluecon.com > [28] mailto:FreeSWITCH-users at lists.freeswitch.org > [29] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [30] http://lists.freeswitch.org/mailman/options/freeswitch-users > [31] http://www.freeswitch.org > [32] mailto:ehermouet at bluetel.fr > [33] mailto:consulting at freeswitch.org > [34] http://www.freeswitchsolutions.com > [35] > [36] http://www.freeswitch.org > [37] http://wiki.freeswitch.org > [38] http://www.cluecon.com > [39] mailto:FreeSWITCH-users at lists.freeswitch.org > [40] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [41] http://lists.freeswitch.org/mailman/options/freeswitch-users > [42] http://www.freeswitch.org > [43] http://pastebin.freeswitch.org/20933 > [44] mailto:ehermouet at bluetel.fr > [45] mailto:msc at freeswitch.org > [46] mailto:ehermouet at bluetel.fr > [47] http://pastebin.freeswitch.org/20927 > [48] mailto:consulting at freeswitch.org > [49] http://www.freeswitchsolutions.com > [50] > [51] http://www.freeswitch.org > [52] http://wiki.freeswitch.org > [53] http://www.cluecon.com > [54] mailto:FreeSWITCH-users at lists.freeswitch.org > [55] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [56] http://lists.freeswitch.org/mailman/options/freeswitch-users > [57] http://www.freeswitch.org > [58] mailto:ehermouet at bluetel.fr > [59] http://pastebin.freeswitch.org/20927 > [60] mailto:msc at freeswitch.org > [61] mailto:consulting at freeswitch.org > [62] http://www.freeswitchsolutions.com > [63] > [64] http://www.freeswitch.org > [65] http://wiki.freeswitch.org > [66] http://www.cluecon.com > [67] mailto:FreeSWITCH-users at lists.freeswitch.org > [68] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [69] http://lists.freeswitch.org/mailman/options/freeswitch-users > [70] http://www.freeswitch.org > [71] mailto:ehermouet at bluetel.fr > [72] mailto:consulting at freeswitch.org > [73] http://www.freeswitchsolutions.com > [74] > [75] http://www.freeswitch.org > [76] http://wiki.freeswitch.org > [77] http://www.cluecon.com > [78] mailto:FreeSWITCH-users at lists.freeswitch.org > [79] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [80] http://lists.freeswitch.org/mailman/options/freeswitch-users > [81] http://www.freeswitch.org > [82] http://wiki.freeswitch.org/wiki/Codec_negotiation > [83] mailto:ehermouet at bluetel.fr > [84] mailto:consulting at freeswitch.org > [85] http://www.freeswitchsolutions.com > [86] > [87] http://www.freeswitch.org > [88] http://wiki.freeswitch.org > [89] http://www.cluecon.com > [90] mailto:FreeSWITCH-users at lists.freeswitch.org > [91] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [92] http://lists.freeswitch.org/mailman/options/freeswitch-users > [93] http://www.freeswitch.org > [94] mailto:ehermouet at bluetel.fr From kbdfck at gmail.com Wed May 22 13:17:30 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Wed, 22 May 2013 13:17:30 +0400 Subject: [Freeswitch-users] Node.JS ESL libraries In-Reply-To: <1369193714184-7590938.post@n2.nabble.com> References: <518D2B52.1090304@digitalmail.com> <1369193714184-7590938.post@n2.nabble.com> Message-ID: https://github.com/englercj/node-esl is modeled after standard FS esl libs and seems to implement full spec I faced some strange troubles with malformed UTF8 symbols in messages, but it works well and is fully async. 2013/5/22 Thomas Lee > Hi, > > Please take a look. > > I'm using node.js and esl (https://github.com/shimaore/esl) > > For example, > -------------------------------------------------- > // > // esl-test1.js > // > > var esl = require('esl'); > var util = require('util'); > > var audio_file = []; > // > // play and get digits > // > > // > audio_file[0]='1 1 3 5000 # sounds/mywav/0.wav'; > audio_file[1]='1 1 3 5000 # sounds/mywav/1.wav'; > audio_file[2]='1 1 3 5000 # sounds/mywav/2.wav'; > audio_file[3]='1 1 3 5000 # sounds/mywav/3.wav'; > audio_file[4]='1 1 3 5000 # sounds/mywav/4.wav'; > audio_file[5]='1 1 3 5000 # sounds/mywav/5.wav'; > audio_file[6]='1 1 3 5000 # sounds/mywav/6.wav'; > audio_file[7]='1 1 3 5000 # sounds/mywav/7.wav'; > audio_file[8]='1 1 3 5000 # sounds/mywav/8.wav'; > audio_file[9]='1 1 3 5000 # sounds/mywav/9.wav'; > audio_file[10]='1 1 3 5000 # sounds/mywav/10.wav'; > audio_file[11]='1 1 3 5000 # sounds/mywav/11.wav'; > audio_file[12]='1 1 3 5000 # sounds/mywav/12.wav'; > audio_file[13]='1 1 3 5000 # sounds/mywav/13.wav'; > > var error3 = '1 1 3 5000 # sounds/mywav/error3.wav'; > var welcome = '1 1 3 5000 # sounds/mywav/hello.wav'; > > > var server = esl.createCallServer(); > //var server = esl.createClient(); > > server.on('CONNECT', function(req, res) { > > var uri, channel_data, unique_id; > > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > > req.execute('answer'); > > //req.execute('playback', 'sounds/mywav/hello.wav'); > > //req.execute('play_and_get_digits', '2 5 3 7000 # > sounds/mywav/hello.wav sounds/mywav/error3.wav myFoo \d+'); > req.execute('play_and_get_digits', welcome); > > req.on('DTMF', function(req) { > > //util.log('DTMF:'+util.inspect(req, null, null)); > > var digit; > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('DTMF: unique_id='+unique_id); > > digit = channel_data['DTMF-Digit']; > console.log('DTMF Received=' + digit); > util.log('DTMF Received'); > if(digit==='#' || digit==='*'){ > > req.execute('play_and_get_digits', error3); > > > } else { > > var n = parseInt(digit); > > req.execute('play_and_get_digits', audio_file[n]); > > > } > > > return; > }); > > req.on('CHANNEL_ANSWER', function(req) { > > //util.log('CHANNEL_ANSWER:'+util.inspect(req, null, > null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_ANSWER: unique_id='+unique_id); > > return util.log('Call was answered'); > }); > > req.on('CHANNEL_HANGUP', function(req) { > > //util.log('CHANNEL_HANGUP:'+util.inspect(req, null, > null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_HANGUP: unique_id='+unique_id); > > console.log('CHANNEL_HANGUP'); > return util.log('CHANNEL_HANGUP'); > }); > > req.on('CHANNEL_HANGUP_COMPLETE', function(req) { > > ///util.log('CHANNEL_HANGUP_COMPLETE:'+util.inspect(req, > null, null)); > > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_HANGUP_COMPLETE: unique_id='+unique_id); > > console.log('CHANNEL_HANGUP_COMPLETE'); > return util.log('CHANNEL_HANGUP_COMPLETE'); > }); > > req.on('DISCONNECT', function(req) { > > //util.log('DISCONNECT:'+util.inspect(req, null, null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('DISCONNECT: unique_id='+unique_id); > > console.log('DISCONNECT'); > return util.log('DISCONNECT'); > }) > > //util.log('CONNECT: req '+util.inspect(req, null, null)); > util.log('CONNECT: unique_id='+unique_id); > > return util.log('CONNECT received'); > > }); > > server.listen(9173); > > ----------------------------------------------- > > Thanks > > Regards, > Thomas Lee > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Node-JS-ESL-libraries-tp7590562p7590938.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://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 > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/3dbca82c/attachment-0001.html From mike at hendrienet.com Wed May 22 15:55:11 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Wed, 22 May 2013 06:55:11 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Thank you. Here is the log URL. http://pastebin.freeswitch.org/20960 On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: > Post a FreeSWITCH debug log of the incoming call. Use > pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax > highlighting. Paste the URL in this email thread and we'll take a look. > -MC > > > On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: > >> Correction: >> I had a second dialplan in the public folder that was causing confusion. >> Below is the dialplan I am using. >> If I change the extension in the dialplan from 1000 to 1001 I get >> the appropriate voice mail extension, however, the phones never ring. >> >> I have the fs configured as a multi-tenant solution. >> >> Could the dialplan be using the default extensions (1000 and 1001) under >> /conf/directory/default and not reference the >> /conf/directory/GothamCity.xom domain? That would explain why I get to the >> voicemail for the correct extension when the phone never rings. >> >> Thanks >> >> ===== >> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >> >> >> >> >> >> >> >> >> >> >> >> >> ===== >> /conf/directory/GothamCity.xom.xml >> >> >> >> >> > value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >> ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ===== >> >> >> >> >> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >> >>> Thank you for your assistance. I made the suggested modification below, >>> however, when calling the number it goes directly to voicemail. >>> >>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/504326ca/attachment.html From philippe at ppmt.org Wed May 22 16:49:43 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Wed, 22 May 2013 08:49:43 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: This is why it goes to voicemail I guess 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[1001]to event 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing variable [sip_invite_domain]=[GothamCity.xom] 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[1001 at GothamCity.xom] 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate Failed. Cause: USER_NOT_REGISTERED 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set Codec sofia/external/5555555555 at 66.66.66.66 P Do you have a dialtone on that 1001 extension? On 22 May 2013 07:55, Mike Hendrie wrote: > Thank you. > > Here is the log URL. > http://pastebin.freeswitch.org/20960 > > > > On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: > >> Post a FreeSWITCH debug log of the incoming call. Use >> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >> highlighting. Paste the URL in this email thread and we'll take a look. >> -MC >> >> >> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >> >>> Correction: >>> I had a second dialplan in the public folder that was causing confusion. >>> Below is the dialplan I am using. >>> If I change the extension in the dialplan from 1000 to 1001 I get >>> the appropriate voice mail extension, however, the phones never ring. >>> >>> I have the fs configured as a multi-tenant solution. >>> >>> Could the dialplan be using the default extensions (1000 and 1001) under >>> /conf/directory/default and not reference the >>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>> voicemail for the correct extension when the phone never rings. >>> >>> Thanks >>> >>> ===== >>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ===== >>> /conf/directory/GothamCity.xom.xml >>> >>> >>> >>> >>> >> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ===== >>> >>> >>> >>> >>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>> >>>> Thank you for your assistance. I made the suggested modification below, >>>> however, when calling the number it goes directly to voicemail. >>>> >>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/056c95a0/attachment-0001.html From mike at hendrienet.com Wed May 22 17:15:50 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Wed, 22 May 2013 08:15:50 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: I do. I can also ring extension 1000. On May 22, 2013 7:52 AM, "Philippe Le Toquin" wrote: > This is why it goes to voicemail I guess > > 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ > 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[1001]to event > 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables > 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing > variable [sip_invite_domain]=[GothamCity.xom] > 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing > variable [presence_id]=[1001 at GothamCity.xom] > 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type > [error] cause: [USER_NOT_REGISTERED] > 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: > 606 [USER_NOT_REGISTERED] > 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type > [user] cause: [USER_NOT_REGISTERED] > 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: > 606 [USER_NOT_REGISTERED] > 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate > Failed. Cause: USER_NOT_REGISTERED > 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() > 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio > Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio > Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set > Codec sofia/external/5555555555 at 66.66.66.66 P > > Do you have a dialtone on that 1001 extension? > > > On 22 May 2013 07:55, Mike Hendrie wrote: > >> Thank you. >> >> Here is the log URL. >> http://pastebin.freeswitch.org/20960 >> >> >> >> On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: >> >>> Post a FreeSWITCH debug log of the incoming call. Use >>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>> highlighting. Paste the URL in this email thread and we'll take a look. >>> -MC >>> >>> >>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>> >>>> Correction: >>>> I had a second dialplan in the public folder that was causing >>>> confusion. Below is the dialplan I am using. >>>> If I change the extension in the dialplan from 1000 to 1001 I get >>>> the appropriate voice mail extension, however, the phones never ring. >>>> >>>> I have the fs configured as a multi-tenant solution. >>>> >>>> Could the dialplan be using the default extensions (1000 and 1001) >>>> under /conf/directory/default and not reference the >>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>> voicemail for the correct extension when the phone never rings. >>>> >>>> Thanks >>>> >>>> ===== >>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ===== >>>> /conf/directory/GothamCity.xom.xml >>>> >>>> >>>> >>>> >>>> >>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ===== >>>> >>>> >>>> >>>> >>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>>> >>>>> Thank you for your assistance. I made the suggested modification >>>>> below, however, when calling the number it goes directly to voicemail. >>>>> >>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>> expression="^1?(262xxxxxxx)$"> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/aeea8f0f/attachment.html From khorsmann at gmail.com Wed May 22 17:53:34 2013 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 22 May 2013 15:53:34 +0200 Subject: [Freeswitch-users] opensips & Freeswitch calling issue In-Reply-To: <1368781652733-7590807.post@n2.nabble.com> References: <1368765501780-7590795.post@n2.nabble.com> <1368781652733-7590807.post@n2.nabble.com> Message-ID: Hi, sorry but i only see the Invites and the register Messages and after then FS returns 408 cause. Maybe you should take an working example of opensips + freeswitch as blueprint for your. < http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration > 2013/5/17 smrdoshi > Hi Karsten, > > Thanks for the reply. > > I took tcpdump of FS sip log. Its showing 408 Request timeout error now. I > am not sure where's the problem? > > I am attaching tcpdump in case if you get idea about the issue. > tcpdump.pcap > > Please let me know. > > Thanks again for your help. > > Thanks, > Sam > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795p7590807.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://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 > -- Mit freundlichen Gr??en *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/76ebf285/attachment-0001.html From mike at hendrienet.com Wed May 22 18:14:36 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Wed, 22 May 2013 09:14:36 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: I will look into the registration of the phones as I see I am missing something. freeswitch at internal> sofia status profile GothamCity.xom reg Registrations: ================================================================================================= Total items returned: 0 ================================================================================================= On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie wrote: > I do. I can also ring extension 1000. > On May 22, 2013 7:52 AM, "Philippe Le Toquin" wrote: > >> This is why it goes to voicemail I guess >> >> 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ >> 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[1001 >> ] to event >> 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables >> 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >> variable [sip_invite_domain]=[GothamCity.xom] >> 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >> variable [presence_id]=[1001 at GothamCity.xom] >> 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >> [error] cause: [USER_NOT_REGISTERED] >> 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >> 606 [USER_NOT_REGISTERED] >> 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >> [user] cause: [USER_NOT_REGISTERED] >> 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >> 606 [USER_NOT_REGISTERED] >> 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate >> Failed. Cause: USER_NOT_REGISTERED >> 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() >> 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio >> Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >> 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663 Audio >> Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set >> Codec sofia/external/5555555555 at 66.66.66.66 P >> >> Do you have a dialtone on that 1001 extension? >> >> >> On 22 May 2013 07:55, Mike Hendrie wrote: >> >>> Thank you. >>> >>> Here is the log URL. >>> http://pastebin.freeswitch.org/20960 >>> >>> >>> >>> On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: >>> >>>> Post a FreeSWITCH debug log of the incoming call. Use >>>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>>> highlighting. Paste the URL in this email thread and we'll take a look. >>>> -MC >>>> >>>> >>>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>>> >>>>> Correction: >>>>> I had a second dialplan in the public folder that was causing >>>>> confusion. Below is the dialplan I am using. >>>>> If I change the extension in the dialplan from 1000 to 1001 I get >>>>> the appropriate voice mail extension, however, the phones never ring. >>>>> >>>>> I have the fs configured as a multi-tenant solution. >>>>> >>>>> Could the dialplan be using the default extensions (1000 and 1001) >>>>> under /conf/directory/default and not reference the >>>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>>> voicemail for the correct extension when the phone never rings. >>>>> >>>>> Thanks >>>>> >>>>> ===== >>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ===== >>>>> /conf/directory/GothamCity.xom.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@ >>>>> ${dialed_domain})}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ===== >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>>>> >>>>>> Thank you for your assistance. I made the suggested modification >>>>>> below, however, when calling the number it goes directly to voicemail. >>>>>> >>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="^1?(262xxxxxxx)$"> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130522/cc9d9b0d/attachment-0001.html From shahzad.bhatti at g-r-v.com Wed May 22 19:00:57 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Wed, 22 May 2013 20:00:57 +0500 Subject: [Freeswitch-users] mod_cdr_csv is not working! Message-ID: Hi, after loading *mod_cdr_csv, *when i originate test calls the cdr is not appended in Master.csv file most of the time but sometime cdr is appended in the Master.csv that show that mod_cdr_csv is configured. Also when i try to use *cdr_csv rotate* command on fs_cli it also not work. even i also try to do using the perl script example available on following url https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_MySQL i want to know why this is happening on my server and how i can fix these issues 1. cdrs are not updated in the csv file; 2. rotate command is not working; any reply is highly appreciated; Regards Shahzad Bhatti -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/b345839e/attachment.html From sdevoy at bizfocused.com Wed May 22 19:07:11 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 22 May 2013 11:07:11 -0400 Subject: [Freeswitch-users] One Way Audio In-Reply-To: References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> <027601ce5689$7de60d20$79b22760$@bizfocused.com> Message-ID: <05df01ce56fe$098b9e70$1ca2db50$@bizfocused.com> Hey Michael, Thanks. I am just throwing this out here for anyone who may stumble on this in archives. My first thought was you just punted! However, I installed wireshark and had a look. I saw enough to figure out what your statement "You are definitely sending RTP" means! FS is send RTP (Real Time Protocol) packets to the carrier. The RTP packets ARE the audio (profound revelation) . So everything is fine with FS and the audio is getting lost between my Carrier (Vitelity) and SPRINT. It is odd that we have had no complaints of lost audio except to SPRINT phones. And I must say I do not have this problem on my SPRINT phone except for calls from my FS. I will post Vitelity's response. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 22, 2013 3:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy wrote: Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n Of course I got 2 way audio. I called the one that ALWAYS fails .. Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy wrote: Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: . I THINK it happens in either direction. . For this person at his home, it appears to be every time (for now) . He reports calling (to or from) other Sprint Cell users results in the same problem from our FS . It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear . he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/511ae6bd/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/511ae6bd/attachment-0001.gif From anthony.minessale at gmail.com Wed May 22 19:12:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 22 May 2013 10:12:02 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: 2.6.25 or newer to get timerfd support. On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: > On 5/22/13, Anthony Minessale wrote: > > You should consider centos6 or debian stable. Make sure the host kernel > is > > very new to get maximum results. > > Tony, do you mean "very new kernel" means 3.2.xx kernel? > > Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not > possible. And that is what CentOS6 offers, too. > > However, I installed FS as openvz guest, it works fine for outgoing, > but not DNAT works for incoming connections even after throroughly > following > http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. > > Just my two cents. > > > > > > > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky > wrote: > > > >> Hello, > >> > >> Do you have any recommendations regarding how to set up correctly (for > >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to > >> tuneup > >> the system to be rock solid? > >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with > FS > >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that > >> something is not well configured around timers, however mod_posix_timer > >> did > >> not help anything (running FS with -rp). We use our own bare metal and > >> can > >> reproduce those delatas eirher when only one VE is on the HW. > >> Maybe time to check out centos6 with openvz? > >> > >> Any idea, recommendation, experience can be very helpful. > >> > >> Regards, > >> Jalsot > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > >> > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/2dca2ad6/attachment.html From lucas_mvoip at hotmail.com Wed May 22 12:46:44 2013 From: lucas_mvoip at hotmail.com (Lucas Tehbing) Date: Wed, 22 May 2013 09:46:44 +0100 Subject: [Freeswitch-users] Routing on the basis of RURI In-Reply-To: References: , Message-ID: Dear Guys, I have been using Opensips as SBC but now want to move to Freeswitch because of Jitter buffer module and many other interesting features. My Call flow is .... SIP UA (Behind NAT) -------------> SBC (Currently opensips)----->Sip Provider (Configured at SIP UA) When SIP message is received at SBC, SBC checks if RURI belongs to some other domain (if(uri==myself) ) if yes it reads the RURI and forwards (State fully) the message (Register, Invite , BYE all kinds of messages). It also modifies SDP in order to be in media path. I want to implement FS in same scenario. The issue is that I cannot configure RURIs on FS, Freeswitch should read from received Request where to forward the message. I am still in the process to learn Freeswitch. If anyone can point me to right resource it will save lot of time. Regards, Lucas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/8d9c5503/attachment-0001.html From lucas_mvoip at hotmail.com Wed May 22 14:45:44 2013 From: lucas_mvoip at hotmail.com (Lucas Tehbing) Date: Wed, 22 May 2013 11:45:44 +0100 Subject: [Freeswitch-users] Routing on the basis of RURI In-Reply-To: References: , , Message-ID: Dear Guys, I have been using Opensips as SBC but now want to move to Freeswitch because of Jitter buffer module and many other interesting features. My Call flow is .... SIP UA (Behind NAT) -------------> SBC (Currently opensips)----->Sip Provider (Configured at SIP UA) When SIP message is received at SBC, SBC checks if RURI belongs to some other domain (if(uri==myself) ) if yes it reads the RURI and forwards (State fully) the message (Register, Invite , BYE all kinds of messages). It also modifies SDP in order to be in media path. I want to implement FS in same scenario. The issue is that I cannot configure RURIs on FS, Freeswitch should read from received Request where to forward the message. I am still in the process to learn Freeswitch. If anyone can point me to right resource it will save lot of time. Regards, Lucas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/deb7ac97/attachment-0001.html From ostolyar at netflix.com Wed May 22 19:31:57 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Wed, 22 May 2013 08:31:57 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy In-Reply-To: References: Message-ID: Clive, Michael, Brian, thank you so much for replying. Please see my answers below. I don't want to shut it down but I need to deploy the system in the cloud where I cannot guarantee that any single instance will stay up. *Clive*, what do you mean by "FreeSWITCH remains in control"? I understand that it remains in control of SIP messages but in the middle of a call there should not be any SIP messages being passed through, so if the proxy server goes down the RTP between the endpoints should keep going. *Michael*, I tried "shutdown now" - it does not seem to work. The documentation says that it will still clean up the current traffic, so it hangs up the call before shutting down. *That being said - if I simply disconnect the proxy machine from the network, the call continues as expected, so in this case it works as I need it to!* *Brian*, we are definitely considering OpenSIPS and other proxies but I'd like to try to set up a system with a single stack to simplify deployment, maintenance, upgrades and such. Thanks again guys! Thank you *Oleg* On Tue, May 21, 2013 at 11:21 AM, Brian Foster wrote: > I'd almost be looking at something like OpenSIPS as FS is a back2back UA, > not a proxy. > On May 21, 2013 2:17 PM, "clive engelberg" wrote: > >> Hi >> >> You can get help in exchange for some netflix movies:) >> >> jk :) >> >> Freeswitch will act as a stateful proxy, meaning it will remain in >> control of the call, even though RTP goes point to point. >> >> Why would you want to be able to shut down FS anyways? >> >> Clive >> >> On Mon, 20 May 2013 19:10:09 -0700 Oleg Stolyar >> wrote >> >> Hi, >> >> I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below >> which simply sets bypass_media to true and then bridges to another >> FreeSWITCH server. >> >> However, when during the call I shut down the proxy FS, the call is >> immediately dropped. Why is that? Is there a way to keep it going? I >> understand that in this case I won't be able to properly send the BYE >> signal when one of the parties hangs up and that's OK. >> >> I tried using redirect and deflect instead of the bridge but those don't >> seem to work at all - probably because my UAs don't know how to handle >> redirects. >> >> >> >> >> >> >> >> >> >> ------------------------------ >> South Africa premier free email service - webmail.co.za >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/a1f98472/attachment-0001.html From msc at freeswitch.org Wed May 22 19:38:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 08:38:41 -0700 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Mike, You might be confusing "profiles" with "domains". A profile is a "SIP profile" and represent a SIP user agent where FreeSWITCH's SIP stack ("Sofia") listens and responds. There is no explicit relationship between domains and profiles. Profiles simply define where SIP messages come and go on the system. Domains are a logical construct in the server. You can have multiple domains and thus multiple users with the same ID, i.e. "1001 at domain1" and "1001 at domain2". Go to your fs_cli and capture the output of these commands: sofia status sofia status profile internal reg Drop those in a pastebin a link back here. -MC On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie wrote: > I will look into the registration of the phones as I see I am missing > something. > > freeswitch at internal> sofia status profile GothamCity.xom reg > > Registrations: > > ================================================================================================= > Total items returned: 0 > > ================================================================================================= > > > > On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie wrote: > >> I do. I can also ring extension 1000. >> On May 22, 2013 7:52 AM, "Philippe Le Toquin" wrote: >> >>> This is why it goes to voicemail I guess >>> >>> 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ >>> 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[ >>> 1001] to event >>> 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables >>> 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >>> variable [sip_invite_domain]=[GothamCity.xom] >>> 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >>> variable [presence_id]=[1001 at GothamCity.xom] >>> 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >>> [error] cause: [USER_NOT_REGISTERED] >>> 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>> 606 [USER_NOT_REGISTERED] >>> 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >>> [user] cause: [USER_NOT_REGISTERED] >>> 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>> 606 [USER_NOT_REGISTERED] >>> 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate >>> Failed. Cause: USER_NOT_REGISTERED >>> 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() >>> 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >>> 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>> 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set >>> Codec sofia/external/5555555555 at 66.66.66.66 P >>> >>> Do you have a dialtone on that 1001 extension? >>> >>> >>> On 22 May 2013 07:55, Mike Hendrie wrote: >>> >>>> Thank you. >>>> >>>> Here is the log URL. >>>> http://pastebin.freeswitch.org/20960 >>>> >>>> >>>> >>>> On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: >>>> >>>>> Post a FreeSWITCH debug log of the incoming call. Use >>>>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>>>> highlighting. Paste the URL in this email thread and we'll take a look. >>>>> -MC >>>>> >>>>> >>>>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>>>> >>>>>> Correction: >>>>>> I had a second dialplan in the public folder that was causing >>>>>> confusion. Below is the dialplan I am using. >>>>>> If I change the extension in the dialplan from 1000 to 1001 I get >>>>>> the appropriate voice mail extension, however, the phones never ring. >>>>>> >>>>>> I have the fs configured as a multi-tenant solution. >>>>>> >>>>>> Could the dialplan be using the default extensions (1000 and 1001) >>>>>> under /conf/directory/default and not reference the >>>>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>>>> voicemail for the correct extension when the phone never rings. >>>>>> >>>>>> Thanks >>>>>> >>>>>> ===== >>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ===== >>>>>> /conf/directory/GothamCity.xom.xml >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@ >>>>>> ${dialed_domain})}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ===== >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>>>>> >>>>>>> Thank you for your assistance. I made the suggested modification >>>>>>> below, however, when calling the number it goes directly to voicemail. >>>>>>> >>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="^1?(262xxxxxxx)$"> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/d209e51f/attachment-0001.html From msc at freeswitch.org Wed May 22 19:39:26 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 08:39:26 -0700 Subject: [Freeswitch-users] mod_cdr_csv is not working! In-Reply-To: References: Message-ID: Can you reproduce this on latest HEAD? If so, gather details and open a Jira. -MC On Wed, May 22, 2013 at 8:00 AM, Shahzad Bhatti wrote: > Hi, > > after loading *mod_cdr_csv, *when i originate test calls the cdr is not > appended in Master.csv file most of the time but sometime cdr is appended > in the Master.csv that show that mod_cdr_csv is configured. Also when i try > to use *cdr_csv rotate* command on fs_cli it also not work. even i also > try to do using the perl script example available on following url > > > https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_MySQL > > i want to know why this is happening on my server and how i can fix these > issues > > 1. cdrs are not updated in the csv file; > 2. rotate command is not working; > > > any reply is highly appreciated; > > Regards > > Shahzad Bhatti > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/6a7c5df9/attachment.html From msc at freeswitch.org Wed May 22 19:44:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 08:44:22 -0700 Subject: [Freeswitch-users] One Way Audio In-Reply-To: <05df01ce56fe$098b9e70$1ca2db50$@bizfocused.com> References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> <027601ce5689$7de60d20$79b22760$@bizfocused.com> <05df01ce56fe$098b9e70$1ca2db50$@bizfocused.com> Message-ID: On Wed, May 22, 2013 at 8:07 AM, Sean Devoy wrote: > Hey Michael,**** > > ** ** > > Thanks.**** > > ** ** > > I am just throwing this out here for anyone who may stumble on this in > archives. > Thanks for sharing your journey along the way. Every new user has this journey and there are occasionally some bumps. You were wise to roll up your sleeves and dive in. That's the best way to learn. While all of this stuff is documented on the wiki and elsewhere on the Web there is simply no substitute for seeing it firsthand. Wireshark is great for that. > **** > > ** ** > > My first thought was you just punted! However, I installed wireshark and > had a look. I saw enough to figure out what your statement ?You are > definitely sending RTP? means! FS is send RTP (Real Time Protocol) > packets to the carrier. The RTP packets ARE the audio (profound > revelation) . So everything is fine with FS and the audio is getting lost > between my Carrier (Vitelity) and SPRINT. > Sometimes the simplest truths are the most profound. ("Is the world round or flat?" or "Is my wife pregnant or not?" :) > **** > > ** ** > > It is odd that we have had no complaints of lost audio except to SPRINT > phones. And I must say I do not have this problem on my SPRINT phone > except for calls from my FS.**** > > ** ** > > I will post Vitelity?s response. > We are eager to hear what they have to say. -MC > **** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Wednesday, May 22, 2013 3:11 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] One Way Audio**** > > ** ** > > You need to show this pcap to your carrier and ask them what's up. You are > definitely sending RTP.**** > > -MC**** > > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/0265f4ac/attachment.html From max at nysolutions.com Wed May 22 19:50:22 2013 From: max at nysolutions.com (Moishe Grunstein) Date: Wed, 22 May 2013 15:50:22 +0000 Subject: [Freeswitch-users] spam>spam>Re: One Way Audio In-Reply-To: <05df01ce56fe$098b9e70$1ca2db50$@bizfocused.com> References: <016401ce5285$27d08cb0$7771a610$@bizfocused.com> <028801ce52a5$086ebe80$194c3b80$@bizfocused.com> <027601ce5689$7de60d20$79b22760$@bizfocused.com> <05df01ce56fe$098b9e70$1ca2db50$@bizfocused.com> Message-ID: Try limiting the codec to just g711u for testing, I remember having something similar with another solution also with Sprint/vitelity before I started using Freeswitch, I don't remember the resolution however I think it was Codec related. 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 Sean Devoy Sent: Wednesday, May 22, 2013 11:07 AM To: 'FreeSWITCH Users Help' Subject: spam>spam>Re: [Freeswitch-users] One Way Audio Hey Michael, Thanks. I am just throwing this out here for anyone who may stumble on this in archives. My first thought was you just punted! However, I installed wireshark and had a look. I saw enough to figure out what your statement "You are definitely sending RTP" means! FS is send RTP (Real Time Protocol) packets to the carrier. The RTP packets ARE the audio (profound revelation) . So everything is fine with FS and the audio is getting lost between my Carrier (Vitelity) and SPRINT. It is odd that we have had no complaints of lost audio except to SPRINT phones. And I must say I do not have this problem on my SPRINT phone except for calls from my FS. I will post Vitelity's response. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, May 22, 2013 3:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio You need to show this pcap to your carrier and ask them what's up. You are definitely sending RTP. -MC On Tue, May 21, 2013 at 6:12 PM, Sean Devoy > wrote: Hey Michael, The problem is back and seems to all the time now. Pcapsipdump for a single call is here http://www.bizfocused.com/sean/fs_problem/pcapsipdump.tar.gz I would like to know why calling FROM FS to a SPRINT phone results in audio FROM SPRINT, but not to SPRINT. Reversing the call works every time. Also, any tips on getting started with wireshark to investigate myself next time would be appreciated. Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, May 17, 2013 7:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio It wouldn't be the first time that a computer decided to behave because it knew Daddy was watching... -MC On Thu, May 16, 2013 at 7:20 PM, Sean Devoy > wrote: Thanks MC. Had to load the pcapdev-lib, but got pcapsipdump installed. My wife had just called my cell and got one way audio. So I ran: pcapsipdump -f -p -i eth0 -d /tmp -n Of course I got 2 way audio. I called the one that ALWAYS fails .... Got 2 way audio! Does pcapsipdump fix it? lol I will try in day time tomorrow and see if we can get a failure. Nothing in the freeswitch.log of value? I didn't see anything, but there is still a lot for me to learn there. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, May 16, 2013 7:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] One Way Audio Sean, Glad to hear you're making progress with using tcpdump and other packet capture-ish tools. You've successfully captured the SIP call leg between your phone and your FreeSWITCH. That's good, but it's incomplete. You really want SIP and RTP both, and you want the call leg between FreeSWITCH and the telco. You have a few options: Expand your tcpdump. In other words, cast a wider net. Pro: easy to do. Con: creates massive pcap files through which you must sift to find the call in question. Use pcapsipdump. Pro: does all the work for you by putting SIP and RTP for each call leg into a single file. Cons: You have to compile it yourself, and it creates a whole lot of files, so you'll need to get used to it. My personal opinion is this: if you never, ever have to debug a SIP call ever again then just use tcpdump. However, if you're the phone guy and you'll be doing this again in the future then bite the bullet and learn pcapsipdump. Believe me it's worth it. -MC On Thu, May 16, 2013 at 3:31 PM, Sean Devoy > wrote: [cid:image003.gif at 01CE56E2.8A2DFB70] Hi all, First, I am on version: FreeSWITCH Version 1.2.5.3+git~20121219T035317Z~2b4aa48049 (git 2b4aa48 2012-12-19 03:53:17Z) I hope to move to the Stable 1.2.9 this weekend. I am having very reliable one way audio when calling Sprint Cell Phone users, though not every time. I got this reproducible on my phone, but by the time I learned tcpdump command, it was working again. However, the user that reported it seems to get it almost everytime. Helpful tidbits: * I THINK it happens in either direction. * For this person at his home, it appears to be every time (for now) * He reports calling (to or from) other Sprint Cell users results in the same problem from our FS * It appears to only be true with Sprint Cell calls! (But my users say that's not Sprints fault!) Scenario: I place a call from my Desk Cisco Phone (220) to his number 410493nnnn and it rings, he answers, I can hear him crystal clear ... he can't hear me at all. I had a theory that it worked after 30 seconds (on my cell), but that does not hold true on his cell. Here is the FS logfile: http://www.bizfocused.com/Sean/fs_problem/freeswitch_no_audio_in.log.txt And here is the tcpdump output: http://www.bizfocused.com/Sean/fs_problem/dump.pcap.zip Based on the small size of the file, I suspect someone is going to say "do it again with this tcpdump command". I welcome the education. Anyway, any insight will be appreciated. Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/6b3ca371/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 2424 bytes Desc: image002.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/6b3ca371/attachment-0001.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.gif Type: image/gif Size: 70 bytes Desc: image003.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/6b3ca371/attachment-0001.gif From alex at digitalmail.com Wed May 22 19:50:34 2013 From: alex at digitalmail.com (Alex Lake) Date: Wed, 22 May 2013 16:50:34 +0100 Subject: [Freeswitch-users] Node.JS ESL libraries In-Reply-To: <1369193714184-7590938.post@n2.nabble.com> References: <518D2B52.1090304@digitalmail.com> <1369193714184-7590938.post@n2.nabble.com> Message-ID: <519CE94A.3020305@digitalmail.com> Oh, that's useful, thanks. > Hi, > > Please take a look. > > I'm using node.js and esl (https://github.com/shimaore/esl) > > For example, > -------------------------------------------------- > // > // esl-test1.js > // > > var esl = require('esl'); > var util = require('util'); > > var audio_file = []; > // > // play and get digits > // > > // > audio_file[0]='1 1 3 5000 # sounds/mywav/0.wav'; > audio_file[1]='1 1 3 5000 # sounds/mywav/1.wav'; > audio_file[2]='1 1 3 5000 # sounds/mywav/2.wav'; > audio_file[3]='1 1 3 5000 # sounds/mywav/3.wav'; > audio_file[4]='1 1 3 5000 # sounds/mywav/4.wav'; > audio_file[5]='1 1 3 5000 # sounds/mywav/5.wav'; > audio_file[6]='1 1 3 5000 # sounds/mywav/6.wav'; > audio_file[7]='1 1 3 5000 # sounds/mywav/7.wav'; > audio_file[8]='1 1 3 5000 # sounds/mywav/8.wav'; > audio_file[9]='1 1 3 5000 # sounds/mywav/9.wav'; > audio_file[10]='1 1 3 5000 # sounds/mywav/10.wav'; > audio_file[11]='1 1 3 5000 # sounds/mywav/11.wav'; > audio_file[12]='1 1 3 5000 # sounds/mywav/12.wav'; > audio_file[13]='1 1 3 5000 # sounds/mywav/13.wav'; > > var error3 = '1 1 3 5000 # sounds/mywav/error3.wav'; > var welcome = '1 1 3 5000 # sounds/mywav/hello.wav'; > > > var server = esl.createCallServer(); > //var server = esl.createClient(); > > server.on('CONNECT', function(req, res) { > > var uri, channel_data, unique_id; > > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > > req.execute('answer'); > > //req.execute('playback', 'sounds/mywav/hello.wav'); > > //req.execute('play_and_get_digits', '2 5 3 7000 # > sounds/mywav/hello.wav sounds/mywav/error3.wav myFoo \d+'); > req.execute('play_and_get_digits', welcome); > > req.on('DTMF', function(req) { > > //util.log('DTMF:'+util.inspect(req, null, null)); > > var digit; > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('DTMF: unique_id='+unique_id); > > digit = channel_data['DTMF-Digit']; > console.log('DTMF Received=' + digit); > util.log('DTMF Received'); > if(digit==='#' || digit==='*'){ > > req.execute('play_and_get_digits', error3); > > > } else { > > var n = parseInt(digit); > > req.execute('play_and_get_digits', audio_file[n]); > > > } > > > return; > }); > > req.on('CHANNEL_ANSWER', function(req) { > > //util.log('CHANNEL_ANSWER:'+util.inspect(req, null, null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_ANSWER: unique_id='+unique_id); > > return util.log('Call was answered'); > }); > > req.on('CHANNEL_HANGUP', function(req) { > > //util.log('CHANNEL_HANGUP:'+util.inspect(req, null, null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_HANGUP: unique_id='+unique_id); > > console.log('CHANNEL_HANGUP'); > return util.log('CHANNEL_HANGUP'); > }); > > req.on('CHANNEL_HANGUP_COMPLETE', function(req) { > > ///util.log('CHANNEL_HANGUP_COMPLETE:'+util.inspect(req, > null, null)); > > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('CHANNEL_HANGUP_COMPLETE: unique_id='+unique_id); > > console.log('CHANNEL_HANGUP_COMPLETE'); > return util.log('CHANNEL_HANGUP_COMPLETE'); > }); > > req.on('DISCONNECT', function(req) { > > //util.log('DISCONNECT:'+util.inspect(req, null, null)); > var channel_data; > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > util.log('DISCONNECT: unique_id='+unique_id); > > console.log('DISCONNECT'); > return util.log('DISCONNECT'); > }) > > //util.log('CONNECT: req '+util.inspect(req, null, null)); > util.log('CONNECT: unique_id='+unique_id); > > return util.log('CONNECT received'); > > }); > > server.listen(9173); > > ----------------------------------------------- > > Thanks > > Regards, > Thomas Lee > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Node-JS-ESL-libraries-tp7590562p7590938.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://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 > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2242 / Virus Database: 3162/5844 - Release Date: 05/21/13 > > From msc at freeswitch.org Wed May 22 20:16:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 09:16:20 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all. Our community conference call is coming up shortly. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22 We don't have a feature presentation today, but we do have several items to discuss so please join us at 1PM EDT, 10AM PDT. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/e5b7436b/attachment.html From ostolyar at netflix.com Wed May 22 22:52:06 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Wed, 22 May 2013 11:52:06 -0700 Subject: [Freeswitch-users] Using FreeSWITCH as a proxy In-Reply-To: References: Message-ID: One correction. Michael, I apologize - you were right "fsctl shutdown now" does indeed leave the call active. The reason it did not work for me is because I just used "shutdown now" and did not see the warning in the documentation that it ignores the arguments in this case. Thank you *Oleg* On Wed, May 22, 2013 at 8:31 AM, Oleg Stolyar wrote: > Clive, Michael, Brian, > thank you so much for replying. Please see my answers below. > > I don't want to shut it down but I need to deploy the system in the cloud > where I cannot guarantee that any single instance will stay up. > *Clive*, what do you mean by "FreeSWITCH remains in control"? I > understand that it remains in control of SIP messages but in the middle of > a call there should not be any SIP messages being passed through, so if the > proxy server goes down the RTP between the endpoints should keep going. > > *Michael*, I tried "shutdown now" - it does not seem to work. The > documentation says that it will still clean up the current traffic, so it > hangs up the call before shutting down. *That being said - if I simply > disconnect the proxy machine from the network, the call continues as > expected, so in this case it works as I need it to!* > > *Brian*, we are definitely considering OpenSIPS and other proxies but I'd > like to try to set up a system with a single stack to simplify deployment, > maintenance, upgrades and such. > > Thanks again guys! > > Thank you > *Oleg* > > > On Tue, May 21, 2013 at 11:21 AM, Brian Foster wrote: > >> I'd almost be looking at something like OpenSIPS as FS is a back2back UA, >> not a proxy. >> On May 21, 2013 2:17 PM, "clive engelberg" wrote: >> >>> Hi >>> >>> You can get help in exchange for some netflix movies:) >>> >>> jk :) >>> >>> Freeswitch will act as a stateful proxy, meaning it will remain in >>> control of the call, even though RTP goes point to point. >>> >>> Why would you want to be able to shut down FS anyways? >>> >>> Clive >>> >>> On Mon, 20 May 2013 19:10:09 -0700 Oleg Stolyar >>> wrote >>> >>> Hi, >>> >>> I am trying to use FreeSWITCH as a SIP proxy. I have the dialplan below >>> which simply sets bypass_media to true and then bridges to another >>> FreeSWITCH server. >>> >>> However, when during the call I shut down the proxy FS, the call is >>> immediately dropped. Why is that? Is there a way to keep it going? I >>> understand that in this case I won't be able to properly send the BYE >>> signal when one of the parties hangs up and that's OK. >>> >>> I tried using redirect and deflect instead of the bridge but those don't >>> seem to work at all - probably because my UAs don't know how to handle >>> redirects. >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ------------------------------ >>> South Africa premier free email service - webmail.co.za >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130522/900e8dc6/attachment.html From info at shishir.com.np Wed May 22 23:38:04 2013 From: info at shishir.com.np (info at shishir.com.np) Date: Wed, 22 May 2013 12:38:04 -0700 Subject: [Freeswitch-users] =?utf-8?q?Mod=5Fdb_deleting_data_from_database?= Message-ID: HI I am setting up the intercept feature, and I went through the wiki on how to accomplish it. The only question I have is once I inserted data on the db with how could I delete this data out of the db ? I see the will delete the record, is this the only options that I have to delete the record from the db ? If yes then how could I use it to delete the record when the call gets hangup? Do I have to have this action on every calls that get hangup? Thanks in advance. From marketing at cluecon.com Wed May 22 23:57:39 2013 From: marketing at cluecon.com (Michael Collins) Date: Wed, 22 May 2013 12:57:39 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: ClueCon Hotel Registration Available Online Message-ID: The ClueCon team would like to announce that the hotel for ClueCon 2013 may now be booked online. Please visit www.ClueCon.com and click the "Book Your Room Online" button. Once you are at the Hyatt site you can enter in your check in and checkout dates and click the Search button. You will be presented with your choice of "standard king" or "standard double" rooms. Keep in mind that the standard double rooms go fast, so don't delay! Be sure to register for ClueCon right away so that we can plan our various activities for when we're all in Chicago. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/10b5cfc6/attachment-0001.html From adahary at gmail.com Thu May 23 00:21:01 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 13:21:01 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server Message-ID: <1369254061700-7590972.post@n2.nabble.com> I would like to integrate FS with RTPproxy like openSips and kamailio are well integrated with it. FS should handle the SIP signaling and the RTPproxy should relay the RTP stream from A to B: A.sip <=> FS <=> B.sip FS = PASS-THRU A.rtp <=> RTPproxy <=> B.rtp. I understand that FS should ask the RTPproxy to allocate UDP ports for both endpoint and then pass-thru-bridge them to cummunicate directly through the RTPproxy. What I cannot yet figure out is how replace both A and B ip/port sets in SDP with the RTPproxy ip/port-udp. I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but couldn't figure it out. how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when when A calls B? Tried the following with no help. Any tip/help/advise will be appriciated. Regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kbdfck at gmail.com Thu May 23 00:38:01 2013 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Thu, 23 May 2013 00:38:01 +0400 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369254061700-7590972.post@n2.nabble.com> References: <1369254061700-7590972.post@n2.nabble.com> Message-ID: What are you trying to achieve with this setup? Centralized nat handling or something else? My thought was that RTPproxy/mediaproxy has to be externally controlled by own protocol to maintain rtp sessions, how do you plan to do that? 2013/5/23 adahary > I would like to integrate FS with RTPproxy like openSips and kamailio are > well integrated with it. > FS should handle the SIP signaling and the RTPproxy should relay the RTP > stream from A to B: > A.sip <=> FS <=> B.sip > FS = PASS-THRU > A.rtp <=> RTPproxy <=> B.rtp. > > I understand that FS should ask the RTPproxy to allocate UDP ports for both > endpoint and then pass-thru-bridge them to cummunicate directly through the > RTPproxy. > > What I cannot yet figure out is how replace both A and B ip/port sets in > SDP > with the RTPproxy ip/port-udp. > I'v read about switch_r_sdp (Leg.A) and switch_m_sdp (Leg.B !?) but > couldn't > figure it out. > > how should I use switch_r_sdp (or/and switch_m_sdp) in my dialplan when > when > A calls B? > Tried the following with no help. > > > > > > Any tip/help/advise will be appriciated. > > Regards > > Assaf > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972.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://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 > -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/2f2874d4/attachment.html From adahary at gmail.com Thu May 23 00:59:09 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 13:59:09 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> Message-ID: <1369256349139-7590974.post@n2.nabble.com> My plan is to use the FS as a centralized SIP proxy/registrar and several RTPproxy media server in different locations so users could get better latency when bridged over. Yes, the RTPproxy is controled by some API commands (google around). I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports and then update A&B SDPs. I'm just missing the last part of 'update A&B SDPs' ; adahary -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590974.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jaybinks at gmail.com Thu May 23 02:47:22 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 23 May 2013 08:47:22 +1000 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: Im using 2.6.32 on all my boxes ... One thing that has me thinking, are there any tweaks to get MSI-X working best it can ? ( with proxmox ) there seems to be a strong bias towards one CPU for all interrupts. I could be wrong, but its something I think ive seen, and didnt see any clear suggestions on. Jay On 23 May 2013 01:12, Anthony Minessale wrote: > 2.6.25 or newer to get timerfd support. > > > > On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: > >> On 5/22/13, Anthony Minessale wrote: >> > You should consider centos6 or debian stable. Make sure the host >> kernel is >> > very new to get maximum results. >> >> Tony, do you mean "very new kernel" means 3.2.xx kernel? >> >> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >> possible. And that is what CentOS6 offers, too. >> >> However, I installed FS as openvz guest, it works fine for outgoing, >> but not DNAT works for incoming connections even after throroughly >> following >> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >> >> Just my two cents. >> >> >> >> > >> > >> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >> wrote: >> > >> >> Hello, >> >> >> >> Do you have any recommendations regarding how to set up correctly (for >> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> >> tuneup >> >> the system to be rock solid? >> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with >> FS >> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> >> something is not well configured around timers, however mod_posix_timer >> >> did >> >> not help anything (running FS with -rp). We use our own bare metal and >> >> can >> >> reproduce those delatas eirher when only one VE is on the HW. >> >> Maybe time to check out centos6 with openvz? >> >> >> >> Any idea, recommendation, experience can be very helpful. >> >> >> >> Regards, >> >> Jalsot >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.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 >> >> >> >> >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/bfe54575/attachment.html From jaybinks at gmail.com Thu May 23 03:18:24 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 23 May 2013 09:18:24 +1000 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369256349139-7590974.post@n2.nabble.com> References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> Message-ID: So why not just to a closer FS node ?? dont get me wrong, I can see some of the appeal for what your suggesting. I have also had similar thoughts in the past... but I keep coming back to the fact you could use FS instead of RTP Proxy... and just sip redirect it. The ONLY downside I could ever see would be... what if the customer dosnt follow the redirect. A re-invite with new IP and port is probably a little safer. Jay On 23 May 2013 06:59, adahary wrote: > My plan is to use the FS as a centralized SIP proxy/registrar and several > RTPproxy media server in different locations so users could get better > latency when bridged over. > > Yes, the RTPproxy is controled by some API commands (google around). > I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports > and then update A&B SDPs. > > I'm just missing the last part of 'update A&B SDPs' ; > > adahary > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590974.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://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 > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/e97cc3fc/attachment-0001.html From rzheng at gmail.com Thu May 23 03:51:03 2013 From: rzheng at gmail.com (Richard Zheng) Date: Wed, 22 May 2013 13:51:03 -1000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: <514C8CB9.2010405@gmail.com> Message-ID: Any resolution on this one? I have the same issue. DTMF coming from a cell phone, say 123, is sometimes repeated, e.g. 11233 when forwarded to an outbound call. It is not consistent though. On Fri, Mar 22, 2013 at 7:16 AM, Matt Broad wrote: > Hi Trever, > > I don't think so, it seems that that jira is for a non-SIP based setup. > > My "issue" is that when calling a PSTN number from a mobile/landline > phone, which gets delivered by SIP to my FreeSwitch server and then routed > out to another PSTN number (via the same provider). When either party > presses a digit, the other end hears the tone. The tone is not the whole > tone just a fraction of it, maybe half a second, but enough to tell the > difference when different keys are pressed. The key can be held for a long > period, say 3 seconds, but you will only hear the "bleeding". > > I have drop_dtmf set to true on both channels (set and export), but I'm > not sure if this would make any difference anyway as the tone is being sent > out-of-band. > > It might also be worth noting that I don't get any duplication of tones. > Pressing 123 will always receive 123 and not 112233. > > thanks > Matt > > > On 22 March 2013 16:54, Trever L. Adams wrote: > >> On 03/12/2013 01:45 PM, Matt Broad wrote: >> > Ok, so the fact that a tone can be heard, although only partially, >> > would suggest inband digits are being sent too. I'll speak to my >> > carrier and see if it is something they can supress their end. >> > Thanks for the help and the tips on debugging :) >> > >> > Thanks >> > Matt >> Matt, by chance is this what you are describing: >> >> http://jira.freeswitch.org/browse/FS-4904 >> >> Trever >> >> > > > -- > Thanks > Matt > > This email and any attachments to it are confidential and are intended > solely for the use of the individual to whom it is addressed. Any views or > opinions expressed are solely those of the author and do not necessarily > represent those of InverOak Limited. > > If you are not the intended recipient of this email, you must neither take > any action based upon its contents, nor copy or show it to anyone. Please > contact the sender if you believe you have received this email in error. > > This email including any attachments cannot be guaranteed to be 100% > secure or error-free as information could be intercepted, corrupted, lost, > destroyed, out-dated, or containing viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message which arise as a result of email transmission. > > InverOak Limited is a company registered in England & Wales under company > number 04529594, whose registered address is Old Barn house, 2 Wannions > Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/82588334/attachment.html From msc at freeswitch.org Thu May 23 04:54:50 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 22 May 2013 17:54:50 -0700 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: <514C8CB9.2010405@gmail.com> Message-ID: Are you using the spandsp dtmf detector by any chance? If not, try it. I think it works a bit better. Look on the dialplan apps section of the wiki for the spands_dtmf* commands. -MC On Wed, May 22, 2013 at 4:51 PM, Richard Zheng wrote: > Any resolution on this one? I have the same issue. DTMF coming from a > cell phone, say 123, is sometimes repeated, e.g. 11233 when forwarded to an > outbound call. It is not consistent though. > > > On Fri, Mar 22, 2013 at 7:16 AM, Matt Broad wrote: > >> Hi Trever, >> >> I don't think so, it seems that that jira is for a non-SIP based setup. >> >> My "issue" is that when calling a PSTN number from a mobile/landline >> phone, which gets delivered by SIP to my FreeSwitch server and then routed >> out to another PSTN number (via the same provider). When either party >> presses a digit, the other end hears the tone. The tone is not the whole >> tone just a fraction of it, maybe half a second, but enough to tell the >> difference when different keys are pressed. The key can be held for a long >> period, say 3 seconds, but you will only hear the "bleeding". >> >> I have drop_dtmf set to true on both channels (set and export), but I'm >> not sure if this would make any difference anyway as the tone is being sent >> out-of-band. >> >> It might also be worth noting that I don't get any duplication of tones. >> Pressing 123 will always receive 123 and not 112233. >> >> thanks >> Matt >> >> >> On 22 March 2013 16:54, Trever L. Adams wrote: >> >>> On 03/12/2013 01:45 PM, Matt Broad wrote: >>> > Ok, so the fact that a tone can be heard, although only partially, >>> > would suggest inband digits are being sent too. I'll speak to my >>> > carrier and see if it is something they can supress their end. >>> > Thanks for the help and the tips on debugging :) >>> > >>> > Thanks >>> > Matt >>> Matt, by chance is this what you are describing: >>> >>> http://jira.freeswitch.org/browse/FS-4904 >>> >>> Trever >>> >>> >> >> >> -- >> Thanks >> Matt >> >> This email and any attachments to it are confidential and are intended >> solely for the use of the individual to whom it is addressed. Any views or >> opinions expressed are solely those of the author and do not necessarily >> represent those of InverOak Limited. >> >> If you are not the intended recipient of this email, you must neither >> take any action based upon its contents, nor copy or show it to anyone. >> Please contact the sender if you believe you have received this email in >> error. >> >> This email including any attachments cannot be guaranteed to be 100% >> secure or error-free as information could be intercepted, corrupted, lost, >> destroyed, out-dated, or containing viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message which arise as a result of email transmission. >> >> InverOak Limited is a company registered in England & Wales under company >> number 04529594, whose registered address is Old Barn house, 2 Wannions >> Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130522/d5177bf5/attachment.html From krice at freeswitch.org Thu May 23 05:02:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 22 May 2013 20:02:08 -0500 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369256349139-7590974.post@n2.nabble.com> Message-ID: Why not just use the front end FS box in bypass media mode, then send the calls to the regional RTP proxies... The sip will act as a controller for those remote locations... Or Use OpenSIPs or Kamilio to handle the front end sip, its designed to use things like RTP Proxy FreeSwitch is not On 5/22/13 3:59 PM, "adahary" wrote: > My plan is to use the FS as a centralized SIP proxy/registrar and several > RTPproxy media server in different locations so users could get better > latency when bridged over. > > Yes, the RTPproxy is controled by some API commands (google around). > I'll use the xml_curl with PHP to request from the RTPproxy ip/udp ports > and then update A&B SDPs. > > I'm just missing the last part of 'update A&B SDPs' ; > > adahary > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media- > server-tp7590972p7590974.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From rzheng at gmail.com Thu May 23 05:27:04 2013 From: rzheng at gmail.com (Richard Zheng) Date: Wed, 22 May 2013 15:27:04 -1000 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: <514C8CB9.2010405@gmail.com> Message-ID: I just use the normal one. Let me try the spandsp one. Thanks, On Wed, May 22, 2013 at 2:54 PM, Michael Collins wrote: > Are you using the spandsp dtmf detector by any chance? If not, try it. I > think it works a bit better. Look on the dialplan apps section of the wiki > for the spands_dtmf* commands. > > -MC > > > On Wed, May 22, 2013 at 4:51 PM, Richard Zheng wrote: > >> Any resolution on this one? I have the same issue. DTMF coming from a >> cell phone, say 123, is sometimes repeated, e.g. 11233 when forwarded to an >> outbound call. It is not consistent though. >> >> >> On Fri, Mar 22, 2013 at 7:16 AM, Matt Broad wrote: >> >>> Hi Trever, >>> >>> I don't think so, it seems that that jira is for a non-SIP based setup. >>> >>> My "issue" is that when calling a PSTN number from a mobile/landline >>> phone, which gets delivered by SIP to my FreeSwitch server and then routed >>> out to another PSTN number (via the same provider). When either party >>> presses a digit, the other end hears the tone. The tone is not the whole >>> tone just a fraction of it, maybe half a second, but enough to tell the >>> difference when different keys are pressed. The key can be held for a long >>> period, say 3 seconds, but you will only hear the "bleeding". >>> >>> I have drop_dtmf set to true on both channels (set and export), but I'm >>> not sure if this would make any difference anyway as the tone is being sent >>> out-of-band. >>> >>> It might also be worth noting that I don't get any duplication of tones. >>> Pressing 123 will always receive 123 and not 112233. >>> >>> thanks >>> Matt >>> >>> >>> On 22 March 2013 16:54, Trever L. Adams wrote: >>> >>>> On 03/12/2013 01:45 PM, Matt Broad wrote: >>>> > Ok, so the fact that a tone can be heard, although only partially, >>>> > would suggest inband digits are being sent too. I'll speak to my >>>> > carrier and see if it is something they can supress their end. >>>> > Thanks for the help and the tips on debugging :) >>>> > >>>> > Thanks >>>> > Matt >>>> Matt, by chance is this what you are describing: >>>> >>>> http://jira.freeswitch.org/browse/FS-4904 >>>> >>>> Trever >>>> >>>> >>> >>> >>> -- >>> Thanks >>> Matt >>> >>> This email and any attachments to it are confidential and are intended >>> solely for the use of the individual to whom it is addressed. Any views or >>> opinions expressed are solely those of the author and do not necessarily >>> represent those of InverOak Limited. >>> >>> If you are not the intended recipient of this email, you must neither >>> take any action based upon its contents, nor copy or show it to anyone. >>> Please contact the sender if you believe you have received this email in >>> error. >>> >>> This email including any attachments cannot be guaranteed to be 100% >>> secure or error-free as information could be intercepted, corrupted, lost, >>> destroyed, out-dated, or containing viruses. The sender therefore does not >>> accept liability for any errors or omissions in the contents of this >>> message which arise as a result of email transmission. >>> >>> InverOak Limited is a company registered in England & Wales under >>> company number 04529594, whose registered address is Old Barn house, 2 >>> Wannions Close, Botley, Chesham, Buckinghamshire, HP5 1YA, United Kingdom. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/d2e4e84b/attachment.html From mike at hendrienet.com Thu May 23 06:48:12 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Wed, 22 May 2013 21:48:12 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <5199827A.5080003@ppmt.org> <519A9A81.7020609@ppmt.org> Message-ID: Sorry it took so long. I updated the log file and your requested details. http://pastebin.freeswitch.org/20963 Thank you On Wed, May 22, 2013 at 10:38 AM, Michael Collins wrote: > Mike, > > You might be confusing "profiles" with "domains". A profile is a "SIP > profile" and represent a SIP user agent where FreeSWITCH's SIP stack > ("Sofia") listens and responds. There is no explicit relationship between > domains and profiles. Profiles simply define where SIP messages come and go > on the system. > > Domains are a logical construct in the server. You can have multiple > domains and thus multiple users with the same ID, i.e. "1001 at domain1" and > "1001 at domain2". > > Go to your fs_cli and capture the output of these commands: > sofia status > sofia status profile internal reg > > Drop those in a pastebin a link back here. > -MC > > > On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie wrote: > >> I will look into the registration of the phones as I see I am missing >> something. >> >> freeswitch at internal> sofia status profile GothamCity.xom reg >> >> Registrations: >> >> ================================================================================================= >> Total items returned: 0 >> >> ================================================================================================= >> >> >> >> On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie wrote: >> >>> I do. I can also ring extension 1000. >>> On May 22, 2013 7:52 AM, "Philippe Le Toquin" >>> wrote: >>> >>>> This is why it goes to voicemail I guess >>>> >>>> 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ >>>> 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[ >>>> 1001] to event >>>> 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables >>>> 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >>>> variable [sip_invite_domain]=[GothamCity.xom] >>>> 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608 Parsing >>>> variable [presence_id]=[1001 at GothamCity.xom] >>>> 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >>>> [error] cause: [USER_NOT_REGISTERED] >>>> 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>>> 606 [USER_NOT_REGISTERED] >>>> 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c:2639Cannot create outgoing channel of type >>>> [user] cause: [USER_NOT_REGISTERED] >>>> 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>>> 606 [USER_NOT_REGISTERED] >>>> 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106 Originate >>>> Failed. Cause: USER_NOT_REGISTERED >>>> 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() >>>> 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >>>> 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>> 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772 Set >>>> Codec sofia/external/5555555555 at 66.66.66.66 P >>>> >>>> Do you have a dialtone on that 1001 extension? >>>> >>>> >>>> On 22 May 2013 07:55, Mike Hendrie wrote: >>>> >>>>> Thank you. >>>>> >>>>> Here is the log URL. >>>>> http://pastebin.freeswitch.org/20960 >>>>> >>>>> >>>>> >>>>> On Wed, May 22, 2013 at 1:39 AM, Michael Collins wrote: >>>>> >>>>>> Post a FreeSWITCH debug log of the incoming call. Use >>>>>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>>>>> highlighting. Paste the URL in this email thread and we'll take a look. >>>>>> -MC >>>>>> >>>>>> >>>>>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>>>>> >>>>>>> Correction: >>>>>>> I had a second dialplan in the public folder that was causing >>>>>>> confusion. Below is the dialplan I am using. >>>>>>> If I change the extension in the dialplan from 1000 to 1001 I get >>>>>>> the appropriate voice mail extension, however, the phones never ring. >>>>>>> >>>>>>> I have the fs configured as a multi-tenant solution. >>>>>>> >>>>>>> Could the dialplan be using the default extensions (1000 and 1001) >>>>>>> under /conf/directory/default and not reference the >>>>>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>>>>> voicemail for the correct extension when the phone never rings. >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> ===== >>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ===== >>>>>>> /conf/directory/GothamCity.xom.xml >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>>>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@ >>>>>>> ${dialed_domain})}"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> value="$${default_areacode}"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> ===== >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>>>>>> >>>>>>>> Thank you for your assistance. I made the suggested modification >>>>>>>> below, however, when calling the number it goes directly to voicemail. >>>>>>>> >>>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> expression="^1?(262xxxxxxx)$"> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>> -- >>>>>> Michael S Collins >>>>>> Twitter: @mercutioviz >>>>>> http://www.FreeSWITCH.org >>>>>> http://www.ClueCon.com >>>>>> http://www.OSTAG.org >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130522/baefa078/attachment-0001.html From adahary at gmail.com Thu May 23 08:38:27 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 21:38:27 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> Message-ID: <1369283907099-7590982.post@n2.nabble.com> Jay, I need that the RTP will flow directly between the endpoints or through a CLEAR relay server. Redirecting SIP won't do it - it is just another new SIP session which will not work PASS-Thru because of the NAT issue. adahary -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590982.html Sent from the freeswitch-users mailing list archive at Nabble.com. From adahary at gmail.com Thu May 23 09:16:25 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 22:16:25 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> Message-ID: <1369286185398-7590983.post@n2.nabble.com> Ken, That's exactly what I'm trying to do! FS bypass media and send to RTPproxy to relay the RTP between the two endpoints. What I'm asking is how to setup FS dialplan to replace the ip/port in both SDP endpoints with the RTPproxy ip/port (given that I already know the RTPproxy ip/port). thanks adahary -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590983.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu May 23 09:23:30 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 23 May 2013 00:23:30 -0500 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369286185398-7590983.post@n2.nabble.com> Message-ID: You miss my point just use FreeSWITCH instead of RTPproxy... On 5/23/13 12:16 AM, "adahary" wrote: > Ken, > > That's exactly what I'm trying to do! FS bypass media and send to RTPproxy > to relay the RTP between the two endpoints. > > What I'm asking is how to setup FS dialplan to replace the ip/port in both > SDP endpoints with the RTPproxy ip/port (given that I already know the > RTPproxy ip/port). > > thanks > > adahary > > > > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media- > server-tp7590972p7590983.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From ehermouet at bluetel.fr Thu May 23 09:27:43 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Thu, 23 May 2013 07:27:43 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> Message-ID: <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> Up please. It very urgent Tks Brian Foster a ?crit?: >No, you will not need to use that. >On May 21, 2013 8:00 AM, wrote: > >> I must use ? if yes where >> and how ? >> >> tks advance for your help >> >> >> Le 2013-05-20 23:11, Brian Foster a ?crit : >> > 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 >mode >> > may not work on a transcoded call. >> > >> > You shouldnt be transcoding if you can help it. Now, Im not sure if >> > that is an empty threat but you should enable late codec >negotiation. >> > Information can be found here: >> > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] >> > >> > -BDF >> > On May 17, 2013 3:48 AM, wrote: >> > >> >> im so stupid :) >> >> tks >> >> >> >> http://pastebin.freeswitch.org/20933 [1] >> >> >> >> called num is 022206... and when i try to use dtmf touch 5 its not >> >> works. >> >> >> >> tks >> >> >> >> Le 2013-05-17 09:25, Ken Rice a ?crit : >> >> > it tells you the password in the popup... this is an anti spam >> >> thing >> >> > >> >> > KenSent from my iPad >> >> > >> >> > On May 17, 2013, at 0:46, Hermouet Erwan > >> [2] [14]> >> >> > wrote: >> >> > >> >> >> On login i try my email...but don t work...i loose here >> >> >> >> >> >> Michael Collins a ?crit : >> >> >> >> >> >>> On Thu, May 16, 2013 at 10:29 AM, Erwan Hermouet >> >> >>> wrote: >> >> >>> >> >> >>>> I have the log but i never found how works pastebin ?? do you >> >> >>>> have tutorial ? >> >> >>> >> >> >>> There isnt a tutorial. You log on, paste your stuff into the >> >> text >> >> >>> box, select FreeSWITCH Log as the syntax highlighting and then >> >> >>> click Send. Copy the URL from the browse address bar. it will >> >> be >> >> >>> something like: >> >> >>> http://pastebin.freeswitch.org/20927 [5] [2] >> >> >>> >> >> >>> -MC >> >> >> >> >> >> Hermouet Erwan >> >> >> Responsable technique >> >> >> Bluetel >> >> > >> >> >> >> >> > >> >> > >> >> >> > >> > >_________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org [6] [4] >> >> >> http://www.freeswitchsolutions.com [7] [5] >> >> >> >> >> >> >> >> >> [8] [6] >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org [9] [7] >> >> >> http://wiki.freeswitch.org [10] [8] >> >> >> http://www.cluecon.com [11] [9] >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org [12] [10] >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> [13] [11] >> >> >> >> >> > >> >> > >> >> >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [14] >> >> >> [12] >> >> >> http://www.freeswitch.org [15] [13] >> >> > >> >> > >> >> > Links: >> >> > ------ >> >> > [1] mailto:ehermouet at bluetel.fr [16] >> >> > [2] http://pastebin.freeswitch.org/20927 [17] >> >> > [3] mailto:msc at freeswitch.org [18] >> >> > [4] mailto:consulting at freeswitch.org [19] >> >> > [5] http://www.freeswitchsolutions.com [20] >> >> > [6] [21] >> >> > [7] http://www.freeswitch.org [22] >> >> > [8] http://wiki.freeswitch.org [23] >> >> > [9] http://www.cluecon.com [24] >> >> > [10] mailto:FreeSWITCH-users at lists.freeswitch.org [25] >> >> > [11] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [26] >> >> > [12] >http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [27] >> >> > [13] http://www.freeswitch.org [28] >> >> > [14] mailto:ehermouet at bluetel.fr [29] >> >> >> >> >> > >> > >_________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org [30] >> >> http://www.freeswitchsolutions.com [31] >> >> >> >> >> >> [32] >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org [33] >> >> http://wiki.freeswitch.org [34] >> >> http://www.cluecon.com [35] >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org [36] >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [37] >> >> >> > >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> [38] >> >> http://www.freeswitch.org [39] >> > >> > >> > Links: >> > ------ >> > [1] http://pastebin.freeswitch.org/20933 >> > [2] mailto:ehermouet at bluetel.fr >> > [3] mailto:msc at freeswitch.org >> > [4] mailto:ehermouet at bluetel.fr >> > [5] http://pastebin.freeswitch.org/20927 >> > [6] mailto:consulting at freeswitch.org >> > [7] http://www.freeswitchsolutions.com >> > [8] >> > [9] http://www.freeswitch.org >> > [10] http://wiki.freeswitch.org >> > [11] http://www.cluecon.com >> > [12] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [13] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [14] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [15] http://www.freeswitch.org >> > [16] mailto:ehermouet at bluetel.fr >> > [17] http://pastebin.freeswitch.org/20927 >> > [18] mailto:msc at freeswitch.org >> > [19] mailto:consulting at freeswitch.org >> > [20] http://www.freeswitchsolutions.com >> > [21] >> > [22] http://www.freeswitch.org >> > [23] http://wiki.freeswitch.org >> > [24] http://www.cluecon.com >> > [25] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [26] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [27] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [28] http://www.freeswitch.org >> > [29] mailto:ehermouet at bluetel.fr >> > [30] mailto:consulting at freeswitch.org >> > [31] http://www.freeswitchsolutions.com >> > [32] >> > [33] http://www.freeswitch.org >> > [34] http://wiki.freeswitch.org >> > [35] http://www.cluecon.com >> > [36] mailto:FreeSWITCH-users at lists.freeswitch.org >> > [37] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > [38] http://lists.freeswitch.org/mailman/options/freeswitch-users >> > [39] http://www.freeswitch.org >> > [40] http://wiki.freeswitch.org/wiki/Codec_negotiation >> > [41] mailto:ehermouet at bluetel.fr >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > >------------------------------------------------------------------------ > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/cf03f7e5/attachment-0001.html From adahary at gmail.com Thu May 23 10:00:12 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 23:00:12 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> <1369286185398-7590983.post@n2.nabble.com> Message-ID: <1369288812268-7590986.post@n2.nabble.com> Ok, how to setup the controller FS (bypass mode) to relay both endpoints RTP through another FS box (acting like RTP proxy = rtpproxy.fs.com)? something like dialplan 'bridge endpointB at rtpproxy.fs.com' ? would that make just the RTP relay between endpoints without initiating a new SIP session with rtpproxy.fs.com? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590986.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu May 23 10:17:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 23 May 2013 01:17:59 -0500 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369288812268-7590986.post@n2.nabble.com> Message-ID: Where you have the following ENDPOINT A <-> FS A <-> FS B <-> FS C <-> ENDPOINT B This is a very lose run down, but FS A and C are in bypass media mode, this causes FS A to simply copy the SDPs from Endpoint A and FS B across the bridge so its not in the media path. Same thing happens on FS C... Leaving media path to go from Endpoint A -> FS B -> Endpoint B Using some tricks with say custom sip headers (sip_h_X- chan vars see the wiki) you can instruct FS B where to send the call from there for final delivery... On 5/23/13 1:00 AM, "adahary" wrote: > Ok, how to setup the controller FS (bypass mode) to relay both endpoints RTP > through another FS box (acting like RTP proxy = rtpproxy.fs.com)? > something like dialplan 'bridge endpointB at rtpproxy.fs.com' ? > would that make just the RTP relay between endpoints without initiating a > new SIP session with rtpproxy.fs.com? > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media- > server-tp7590972p7590986.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From adahary at gmail.com Thu May 23 10:52:16 2013 From: adahary at gmail.com (adahary) Date: Wed, 22 May 2013 23:52:16 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> <1369286185398-7590983.post@n2.nabble.com> <1369288812268-7590986.post@n2.nabble.com> Message-ID: <1369291936118-7590988.post@n2.nabble.com> Are FS.A and FS.C logically the same FS box which where both ENDPOINTs A&B registered to? ENDPOINT A <-> FS A <-> ENDPOINT B (sip registration) after FS A bypass mode and 'bridge ENDPOINT.B at FS.B' ENDPOINT.A <-> FS B <-> ENDPOINT.B (rtp relay) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590988.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Thu May 23 10:59:42 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 23 May 2013 01:59:42 -0500 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369291936118-7590988.post@n2.nabble.com> Message-ID: They may or may not be the same box. On 5/23/13 1:52 AM, "adahary" wrote: > Are FS.A and FS.C logically the same FS box which where both ENDPOINTs A&B > registered to? > > ENDPOINT A <-> FS A <-> ENDPOINT B (sip registration) > after FS A bypass mode and 'bridge ENDPOINT.B at FS.B' > ENDPOINT.A <-> FS B <-> ENDPOINT.B (rtp relay) > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media- > server-tp7590972p7590988.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From nandy1925 at gmail.com Thu May 23 11:19:51 2013 From: nandy1925 at gmail.com (nandy) Date: Thu, 23 May 2013 00:19:51 -0700 (PDT) Subject: [Freeswitch-users] Newbie's installation: Can't start FS at boot In-Reply-To: References: Message-ID: <1369293591066-7590990.post@n2.nabble.com> The service script runs Freeswitch as "freeswitch" user. So it fails to run and left the lock file intact. Just execute this command to change the ownership of Freeswitch files: # find /usr/local/freeswitch -exec chown freeswitch:freeswitch {} \; -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Newbie-s-installation-Can-t-start-FS-at-boot-tp2432544p7590990.html Sent from the freeswitch-users mailing list archive at Nabble.com. From adahary at gmail.com Thu May 23 11:26:48 2013 From: adahary at gmail.com (adahary) Date: Thu, 23 May 2013 00:26:48 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: References: <1369254061700-7590972.post@n2.nabble.com> <1369256349139-7590974.post@n2.nabble.com> <1369286185398-7590983.post@n2.nabble.com> <1369288812268-7590986.post@n2.nabble.com> <1369291936118-7590988.post@n2.nabble.com> Message-ID: <1369294008290-7590991.post@n2.nabble.com> Ok, this is what I'm about to setup for testing your solution: 1. registering ENDPOINT.1 & 2 (both behind NAT) with FS.A setup to bypass mode. FS.A has dialplan of ENDPOINT.A at FS.B 2. Setup FS.B seperatly (not in the SIP/media path of ENDPOINT.1 & 2) 3. Calling from ENDPOINT.1 to ENDPOINT.2 4. The Call hits the FS.A dialplan and being bridged over to FS.B (ENDPOINT.A at FS.B) Now, on FS.B I should see (wireshark) only RTP UDP stream between ENDPOINT.1 & 2 with NO SIP (re-direct, re-invite). Is it right? Please be noticed that: - Both ENDPOINT.1 & 2 are beind NAT - ZRTP may be supported for end2end (another reason to have clear RTP relay). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590991.html Sent from the freeswitch-users mailing list archive at Nabble.com. From netcentrica at gmail.com Thu May 23 12:11:57 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 10:11:57 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg Message-ID: Hi, I have very strange problem with audio after upgrading of FS to the newest GIT version. Previous version was quite old, but worked without problems for a long time: FreeSWITCH version: 1.0.head (git-313b164 2011-11-26 08-53-01 -0600) Current version is: FreeSWITCH version: 1.5.1b () All configuration files are the same (exact copy) System: Debian Lenny 32-bit All calls are made over SIP Symptoms: - IVR prompt playback is OK - FAX sending/receiving is OK - Calling to remote servers via VoIP provider is OK - The only problem is calling to local user and it's very strange. When external user calls local user (SIP call, via IVR, then bridge), audio stream from external user is OK, but audio stream from local user sent to external user is very delayed. Delay increases during a call. At the beginning is about 1 second, all the time delay is increasing. Audio is reaching external user but after about 30 seconds of call stream is delayed for about 10 seconds. At the same time audio stream from external user is heard by local user with no delays. Propably something changed in FS core or config files and I need to adjust to the new version. As I stated before configuration is exact copy and worked perfectly on older version. Any hints what should I check or change to solve this problem? Kind Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/ae90d401/attachment.html From andpe at poczta.onet.pl Thu May 23 12:17:19 2013 From: andpe at poczta.onet.pl (andpe) Date: Thu, 23 May 2013 10:17:19 +0200 Subject: [Freeswitch-users] The number of registered phones Message-ID: <16170648-aef23a1b347490dd3b9c62c0a1a0bdd2@pmq2v.m5r2.onet> Hi What is the possible number of registered phones to FreeSWITCH? I know it may depend on many factors. Where can I find this information? Is 2000 or 3000 users a lot or not? I'm not asking about the performance of connections here. Regards Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/71ba9f00/attachment.html From alex at opensystems.net.au Thu May 23 12:32:55 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Thu, 23 May 2013 16:32:55 +0800 Subject: [Freeswitch-users] The number of registered phones In-Reply-To: <16170648-aef23a1b347490dd3b9c62c0a1a0bdd2@pmq2v.m5r2.onet> References: <16170648-aef23a1b347490dd3b9c62c0a1a0bdd2@pmq2v.m5r2.onet> Message-ID: I know some ISP's use freeswitch for their VOIP offerings so I'd guess 2000-3000 isn't a problem. Not sure what type of spec's a system woul need though. *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 23 May 2013 16:17, andpe wrote: > Hi > What is the possible number of registered phones to FreeSWITCH? I know it may > depend on many factors. Where can I find this information? Is 2000 or 3000 > users a lot or not? I'm not asking about the performance of connections > here. > > Regards > Andy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/df9d1b4c/attachment-0001.html From netcentrica at gmail.com Thu May 23 12:40:07 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 10:40:07 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: Additional info from FS console: 2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/myusername at my.office_pbx.ip [BREAK] 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal sofia/external/caller_number at my.fs.ip [BREAK] 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 480 bytes to accommodate 320->480 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging Write Buffer at 320 bytes to accommodate 320->320 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 Asynchronous PTIME not supported, changing our end from 0 to 20 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already using PCMA -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/9cde9a30/attachment.html From adahary at gmail.com Thu May 23 13:13:11 2013 From: adahary at gmail.com (adahary) Date: Thu, 23 May 2013 02:13:11 -0700 (PDT) Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369294008290-7590991.post@n2.nabble.com> References: <1369256349139-7590974.post@n2.nabble.com> <1369286185398-7590983.post@n2.nabble.com> <1369288812268-7590986.post@n2.nabble.com> <1369291936118-7590988.post@n2.nabble.com> <1369294008290-7590991.post@n2.nabble.com> Message-ID: <1369300391544-7590996.post@n2.nabble.com> Ken, It does not work for me. please check the log below. ENDPOINT.1 = 1001 ENDPOINT.2 = 1000 2013-05-23 13:10:01.984411 [INFO] mod_dialplan_xml.c:558 Processing 1001 <1001>->1000 in context confitele_endpoints 2013-05-23 13:10:02.004372 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/42341cdc-9982-4845-88be-0576fbd60ea7.tmp.xml Dialplan: sofia/endpoint-nat/1001 at fs-a.com parsing [confitele_endpoints->call_inbound] continue=false Dialplan: sofia/endpoint-nat/1001 at fs-a.com Regex (PASS) [call_inbound] destination_number(1000) =~ /^(\d+)$/ break=on-false Dialplan: sofia/endpoint-nat/1001 at fs-a.com Action set(bypass_media=true) INLINE EXECUTE sofia/endpoint-nat/1001 at fs-a.com set(bypass_media=true) 2013-05-23 13:10:02.004372 [DEBUG] mod_dptools.c:1373 sofia/endpoint-nat/1001 at fs-a.com SET [bypass_media]=[true] Dialplan: sofia/endpoint-nat/1001 at fs-a.com Action bridge(${destination_number}@fs-b.com) 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:167 (sofia/endpoint-nat/1001 at fs-a.com) State Change CS_ROUTING -> CS_EXECUTE 2013-05-23 13:10:02.004372 [DEBUG] switch_core_session.c:1340 Send signal sofia/endpoint-nat/1001 at fs-a.com [BREAK] 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:470 (sofia/endpoint-nat/1001 at fs-a.com) State ROUTING going to sleep 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:415 (sofia/endpoint-nat/1001 at fs-a.com) Running State Change CS_EXECUTE 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:477 (sofia/endpoint-nat/1001 at fs-a.com) State EXECUTE 2013-05-23 13:10:02.004372 [DEBUG] mod_sofia.c:230 sofia/endpoint-nat/1001 at fs-a.com SOFIA EXECUTE 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:209 sofia/endpoint-nat/1001 at fs-a.com Standard EXECUTE EXECUTE sofia/endpoint-nat/1001 at fs-a.com bridge(1000 at fs-b.com) 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:2039 Parsing global variables 2013-05-23 13:10:02.004372 [ERR] switch_core_session.c:496 Could not locate channel type 1000 at fs-b.com 2013-05-23 13:10:02.004372 [NOTICE] switch_ivr_originate.c:2649 Cannot create outgoing channel of type [1000 at fs-b.com] cause: [CHAN_NOT_IMPLEMENTED] 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:3615 Originate Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] 2013-05-23 13:10:02.004372 [INFO] mod_dptools.c:3098 Originate Failed. Cause: CHAN_NOT_IMPLEMENTED 2013-05-23 13:10:02.004372 [NOTICE] mod_dptools.c:3218 Hangup sofia/endpoint-nat/1001 at fs-a.com [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED] -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media-server-tp7590972p7590996.html Sent from the freeswitch-users mailing list archive at Nabble.com. From acrow at integrafin.co.uk Thu May 23 13:28:27 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 23 May 2013 10:28:27 +0100 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: <519DE13B.8060301@integrafin.co.uk> Probably best to open a bug on jira.freeswitch.org. On 23/05/13 09:40, Adam Raszynski wrote: > Additional info from FS console: > > 2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA > 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send > signal sofia/external/myusername at my.office_pbx.ip [BREAK] > 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send > signal sofia/external/caller_number at my.fs.ip [BREAK] > 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging > Write Buffer at 480 bytes to accommodate 320->480 > 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging > Write Buffer at 320 bytes to accommodate 320->320 > 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 > Asynchronous PTIME not supported, changing our end from 0 to 20 > 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already > using PCMA > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/e26a81b3/attachment.html From jaybinks at gmail.com Thu May 23 13:44:16 2013 From: jaybinks at gmail.com (jay binks) Date: Thu, 23 May 2013 19:44:16 +1000 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: I wonder if this is related to your Asynchronous PTIME ??? are you doing 20ms in one direction and 30 the other ?? On 23 May 2013 18:40, Adam Raszynski wrote: > Additional info from FS console: > > 2013-05-23 10:30:40.402324 [DEBUG] mod_sofia.c:642 SOFIA EXCHANGE_MEDIA > 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal > sofia/external/myusername at my.office_pbx.ip [BREAK] > 2013-05-23 10:30:40.402324 [DEBUG] switch_core_session.c:923 Send signal > sofia/external/caller_number at my.fs.ip [BREAK] > 2013-05-23 10:30:40.422955 [DEBUG] switch_core_io.c:1365 Engaging Write > Buffer at 480 bytes to accommodate 320->480 > 2013-05-23 10:30:40.450958 [DEBUG] switch_core_io.c:1365 Engaging Write > Buffer at 320 bytes to accommodate 320->320 > 2013-05-23 10:30:40.662295 [WARNING] switch_core_media.c:1370 Asynchronous > PTIME not supported, changing our end from 0 to 20 > 2013-05-23 10:30:40.662295 [DEBUG] switch_core_media.c:1741 Already using > PCMA > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/123b43cc/attachment.html From sertys at gmail.com Thu May 23 13:44:39 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 23 May 2013 11:44:39 +0200 Subject: [Freeswitch-users] The number of registered phones In-Reply-To: <16170648-aef23a1b347490dd3b9c62c0a1a0bdd2@pmq2v.m5r2.onet> References: <16170648-aef23a1b347490dd3b9c62c0a1a0bdd2@pmq2v.m5r2.onet> Message-ID: Registrations are by no way expensive on FS. You can easily reach the 10k threshold on a normal dedicated server. If you set a big enough session expiration and use tcp, you can reach way higher. This is given your peers don't make any calls... On May 23, 2013 11:25 AM, "andpe" wrote: > Hi > What is the possible number of registered phones to FreeSWITCH? I know it may > depend on many factors. Where can I find this information? Is 2000 or 3000 > users a lot or not? I'm not asking about the performance of connections > here. > > Regards > Andy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/c8af471a/attachment.html From netcentrica at gmail.com Thu May 23 14:11:22 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 12:11:22 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: I use default sofia settings. I'm not sure how can I "do" or not to do 20 or 30ms - what params are responsible for that? For the calling party I have no control - call comes from external provider, but I can change this for local user. What params should I change? 2013/5/23 jay binks > I wonder if this is related to your Asynchronous PTIME ??? > are you doing 20ms in one direction and 30 the other ?? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/fe818045/attachment-0001.html From netcentrica at gmail.com Thu May 23 14:24:40 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 12:24:40 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: I changed outbound-codec-prefs to PCMU at 20i,PCMA at 20i so timing on inbound and outbound legs is the same Info about async PTIME does not appear now in console, so I think now there are synchronous But the problem is not solved - audio stream from local user is still lagging -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/aa88f5e9/attachment.html From netcentrica at gmail.com Thu May 23 14:51:49 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 12:51:49 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: Changed also rtp-timer-name to none now bridged call is OK, but when local user uses IVR provided by bind-meta-app it gets distorted audio, when leaving IVR and switching back to the caller audio is OK again 2013/5/23 Adam Raszynski > I changed outbound-codec-prefs to PCMU at 20i,PCMA at 20i so timing on inbound > and outbound legs is the same > > Info about async PTIME does not appear now in console, so I think now > there are synchronous > > But the problem is not solved - audio stream from local user is still > lagging > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/f60ff780/attachment.html From philippe at ppmt.org Thu May 23 15:12:57 2013 From: philippe at ppmt.org (Philippe Le Toquin) Date: Thu, 23 May 2013 07:12:57 -0400 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <519A9A81.7020609@ppmt.org> Message-ID: <519DF9B9.5010506@ppmt.org> so your extension are not registered On my FS I can see my own extension. Now this is where I am going to learn something because I have no idea why you can call from one extension to another if they are not registered. On 13-05-22 10:48 PM, Mike Hendrie wrote: > Sorry it took so long. > > I updated the log file and your requested details. > > http://pastebin.freeswitch.org/20963 > > Thank you > > > On Wed, May 22, 2013 at 10:38 AM, Michael Collins > wrote: > > Mike, > > You might be confusing "profiles" with "domains". A profile is a > "SIP profile" and represent a SIP user agent where FreeSWITCH's > SIP stack ("Sofia") listens and responds. There is no explicit > relationship between domains and profiles. Profiles simply define > where SIP messages come and go on the system. > > Domains are a logical construct in the server. You can have > multiple domains and thus multiple users with the same ID, i.e. > "1001 at domain1" and "1001 at domain2". > > Go to your fs_cli and capture the output of these commands: > sofia status > sofia status profile internal reg > > Drop those in a pastebin a link back here. > -MC > > > On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie > wrote: > > I will look into the registration of the phones as I see I am > missing something. > > freeswitch at internal> sofia status profile GothamCity.xom reg > > Registrations: > ================================================================================================= > Total items returned: 0 > ================================================================================================= > > > > On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie > > wrote: > > I do. I can also ring extension 1000. > > On May 22, 2013 7:52 AM, "Philippe Le Toquin" > > wrote: > > This is why it goes to voicemail I guess > > 1. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_channel.c:1099 > sofia/external/5555555555 at 66.66.66.66 > EXPORTING[export_vars] [dialed_extension]=[1001] > to event > 2. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_ivr_originate.c:2044 Parsing global variables > 3. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_event.c:1608 Parsing variable > [sip_invite_domain]=[GothamCity.xom] > 4. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_event.c:1608 Parsing variable > [presence_id]=[1001 at GothamCity.xom] > 5. > 2013-05-21 22:28:54.107400 [NOTICE] > switch_ivr_originate.c:2639 Cannot create outgoing > channel of type [error] cause: [USER_NOT_REGISTERED] > 6. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_ivr_originate.c:3605 Originate Resulted in > Error Cause: 606 [USER_NOT_REGISTERED] > 7. > 2013-05-21 22:28:54.107400 [NOTICE] > switch_ivr_originate.c:2639 Cannot create outgoing > channel of type [user] cause: [USER_NOT_REGISTERED] > 8. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_ivr_originate.c:3605 Originate Resulted in > Error Cause: 606 [USER_NOT_REGISTERED] > 9. > 2013-05-21 22:28:54.107400 [INFO] > mod_dptools.c:3106 Originate Failed. Cause: > USER_NOT_REGISTERED > 10. > EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() > 11. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_core_media.c:2663 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 12. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_core_media.c:2663 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 13. > 2013-05-21 22:28:54.107400 [DEBUG] > switch_core_media.c:1772 Set Codec > sofia/external/5555555555 at 66.66.66.66 P > > Do you have a dialtone on that 1001 extension? > > > > On 22 May 2013 07:55, Mike Hendrie > > wrote: > > Thank you. > > Here is the log URL. > http://pastebin.freeswitch.org/20960 > > > > On Wed, May 22, 2013 at 1:39 AM, Michael Collins > > > wrote: > > Post a FreeSWITCH debug log of the incoming > call. Use pastebin.freeswitch.org > and select > "FreeSWITCH Log" as the syntax highlighting. > Paste the URL in this email thread and we'll > take a look. > -MC > > > On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie > > wrote: > > Correction: > I had a second dialplan in the public > folder that was causing confusion. Below > is the dialplan I am using. > If I change the extension in the dialplan > from 1000 to 1001 I get > the appropriate voice mail extension, > however, the phones never ring. > > I have the fs configured as a > multi-tenant solution. > > Could the dialplan be using the default > extensions (1000 and 1001) under > /conf/directory/default and not reference > the /conf/directory/GothamCity.xom domain? > That would explain why I get to the > voicemail for the correct extension when > the phone never rings. > > Thanks > > ===== > /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml > > > > > expression="^1?(2624481175 > )$"> > data="domain_name=GothamCity.xom"/> > > > > > > > ===== > /conf/directory/GothamCity.xom.xml > > > > > value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/> > > > > > value="default"/> > value="$${default_provider}"/> > value="$${default_areacode}"/> > name="transfer_fallback_extension" > value="operator"/> > > > > > > data="GothamCity.xom/*.xml"/> > > > > > > > > > > > > > > ===== > > > > > On Tue, May 21, 2013 at 9:27 PM, Mike > Hendrie > wrote: > > Thank you for your assistance. I made > the suggested modification below, > however, when calling the number it > goes directly to voicemail. > > /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml > > > > > expression="^1?(262xxxxxxx)$"> > data="1000 XML default"/> > > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/d72ad74c/attachment-0001.html From ashish at nms.co.in Thu May 23 15:27:39 2013 From: ashish at nms.co.in (Ashish gautam) Date: Thu, 23 May 2013 16:57:39 +0530 Subject: [Freeswitch-users] Max. Number of 8 span PRI cards Message-ID: Hi, I want to ask that is it possible to use 3 eight span PRI cards on a single FS server? I mean is there any limit up to which the system performs optimally on this? Please throw some light on this. Thanks -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd. +91 8802865008 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/0adb289c/attachment.html From jos at firstcom.dk Thu May 23 15:36:46 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Thu, 23 May 2013 13:36:46 +0200 Subject: [Freeswitch-users] Lua creating multiple session's Message-ID: Hello List, Im trying to do dial multiple destination through lua, on a single incoming call. I do know that i could do a simple session:execute("bridge", "dst1,dst2,dst3") but I need to do it in individual session, for processing I need to do in a later version of the lua script. I loop through the destinations that needs to be called, creating a new session for each destination, and storing that in an array. Firstly, it seems as freeswitch.Session doesn't reply right away, but waits for early-media. Thats ok though, but makes dialling mobile devices a rather long wait. The problem is, that when I do the second freeswitch.Session, it seems to hold up further lua processing, until the last created call is answered. Is this how its supposed to work. Heres my current code: local legs = {} for key, dev in pairs(dstDevices) do freeswitch.consoleLog("info", "Lets call " .. dev.username .. " with tech " .. dev.devicetech .. "\n") if dev.devicetech == "1" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on uasbc\n") legs[key] = freeswitch.Session("sofia/gateway/uasbc01/" .. dev.username); freeswitch.consoleLog("info", "called " .. dev.username .. " on uasbc\n") elseif dev.devicetech == "2" then freeswitch.consoleLog("info", "calling " .. dev.username .. " on ccsbc, mvno\n") legs[key] = freeswitch.Session("{origination_caller_id_name=+xxxxxxxx,origination_calle r_id_number=+xxxxxxxxx}sofia/gateway/ccsbc01/+xx" .. string.match(dev.username, "^mvno_(.+)")) freeswitch.consoleLog("info", "called " .. dev.username .. " on ccsbc, mvno\n") end End freeswitch.consoleLog("info", "Enter loop now\n") It doesn't reach the last consoleLog until the last call is answered. Venlig hilsen/kind regards Jon Leren Sch?pzinsky From mike at hendrienet.com Thu May 23 15:45:06 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Thu, 23 May 2013 06:45:06 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: <519DF9B9.5010506@ppmt.org> References: <519A9A81.7020609@ppmt.org> <519DF9B9.5010506@ppmt.org> Message-ID: I made a change to 1000 this morning and I can now dial each others extensions. When I call the number I still go directly to the voicemail when extension 1000 should ring. freeswitch at internal> sofia status profile internal reg Registrations: ================================================================================================= Call-ID: 519830183003 at 10.2.1.209 User: 1001 at gothamcity.xom Contact: "user" Agent: Sipdroid/3.0 beta/SCH-I605 Status: Registered(UDP-NAT)(unknown) EXP(2013-05-22 23:44:29) EXPSECS(3501) Host: GothamCity-00 IP: 10.2.1.209 Port: 45048 Auth-User: 1001 Auth-Realm: gothamcity.xom MWI-Account: 1001 at gothamcity.xom Call-ID: 7eb5078e-316ad43b-c274631c at 10.2.1.50 User: 1000 at gothamcity.xom Contact: "user" Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Status: Registered(TCP-NAT)(unknown) EXP(2013-05-22 22:48:44) EXPSECS(156) Host: GothamCity-00 IP: 10.2.1.50 Port: 64461 Auth-User: 1000 Auth-Realm: gothamcity.xom MWI-Account: 1000 at gothamcity.xom Total items returned: 2 ================================================================================================= Mike Hendrie T: 847.366.5881 E: mike at hendrienet.com On May 23, 2013 6:16 AM, "Philippe Le Toquin" wrote: > so your extension are not registered > > On my FS I can see my own extension. > > Now this is where I am going to learn something because I have no idea why > you can call from one extension to another if they are not registered. > > On 13-05-22 10:48 PM, Mike Hendrie wrote: > > Sorry it took so long. > > I updated the log file and your requested details. > > http://pastebin.freeswitch.org/20963 > > Thank you > > > On Wed, May 22, 2013 at 10:38 AM, Michael Collins wrote: > >> Mike, >> >> You might be confusing "profiles" with "domains". A profile is a "SIP >> profile" and represent a SIP user agent where FreeSWITCH's SIP stack >> ("Sofia") listens and responds. There is no explicit relationship between >> domains and profiles. Profiles simply define where SIP messages come and go >> on the system. >> >> Domains are a logical construct in the server. You can have multiple >> domains and thus multiple users with the same ID, i.e. "1001 at domain1" >> and "1001 at domain2". >> >> Go to your fs_cli and capture the output of these commands: >> sofia status >> sofia status profile internal reg >> >> Drop those in a pastebin a link back here. >> -MC >> >> >> On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie wrote: >> >>> I will look into the registration of the phones as I see I am missing >>> something. >>> >>> freeswitch at internal> sofia status profile GothamCity.xom reg >>> >>> Registrations: >>> >>> ================================================================================================= >>> Total items returned: 0 >>> >>> ================================================================================================= >>> >>> >>> >>> On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie wrote: >>> >>>> I do. I can also ring extension 1000. >>>> On May 22, 2013 7:52 AM, "Philippe Le Toquin" >>>> wrote: >>>> >>>>> This is why it goes to voicemail I guess >>>>> >>>>> 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ >>>>> 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[ >>>>> 1001] to event >>>>> 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:2044Parsing global variables >>>>> 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608Parsing variable >>>>> [sip_invite_domain]=[GothamCity.xom] >>>>> 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608Parsing variable >>>>> [presence_id]=[1001 at GothamCity.xom] >>>>> 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c: >>>>> 2639 Cannot create outgoing channel of type [error] cause: [ >>>>> USER_NOT_REGISTERED] >>>>> 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>>>> 606 [USER_NOT_REGISTERED] >>>>> 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c: >>>>> 2639 Cannot create outgoing channel of type [user] cause: [ >>>>> USER_NOT_REGISTERED] >>>>> 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c:3605Originate Resulted in Error Cause: >>>>> 606 [USER_NOT_REGISTERED] >>>>> 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106Originate Failed. Cause: USER_NOT_REGISTERED >>>>> 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() >>>>> 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >>>>> 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>>> 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772Set Codec sofia/external/ >>>>> 5555555555 at 66.66.66.66 P >>>>> >>>>> Do you have a dialtone on that 1001 extension? >>>>> >>>>> >>>>> On 22 May 2013 07:55, Mike Hendrie wrote: >>>>> >>>>>> Thank you. >>>>>> >>>>>> Here is the log URL. >>>>>> http://pastebin.freeswitch.org/20960 >>>>>> >>>>>> >>>>>> >>>>>> On Wed, May 22, 2013 at 1:39 AM, Michael Collins >>>>> > wrote: >>>>>> >>>>>>> Post a FreeSWITCH debug log of the incoming call. Use >>>>>>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>>>>>> highlighting. Paste the URL in this email thread and we'll take a look. >>>>>>> -MC >>>>>>> >>>>>>> >>>>>>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>>>>>> >>>>>>>> Correction: >>>>>>>> I had a second dialplan in the public folder that was causing >>>>>>>> confusion. Below is the dialplan I am using. >>>>>>>> If I change the extension in the dialplan from 1000 to 1001 I get >>>>>>>> the appropriate voice mail extension, however, the phones never ring. >>>>>>>> >>>>>>>> I have the fs configured as a multi-tenant solution. >>>>>>>> >>>>>>>> Could the dialplan be using the default extensions (1000 and >>>>>>>> 1001) under /conf/directory/default and not reference the >>>>>>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>>>>>> voicemail for the correct extension when the phone never rings. >>>>>>>> >>>>>>>> Thanks >>>>>>>> >>>>>>>> ===== >>>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ===== >>>>>>>> /conf/directory/GothamCity.xom.xml >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>>>>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@ >>>>>>>> ${dialed_domain})}"/> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> value="$${default_provider}"/> >>>>>>>> >>>>>>> value="$${default_areacode}"/> >>>>>>>> >>>>>>> value="operator"/> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> ===== >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie wrote: >>>>>>>> >>>>>>>>> Thank you for your assistance. I made the suggested modification >>>>>>>>> below, however, when calling the number it goes directly to voicemail. >>>>>>>>> >>>>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> expression="^1?(262xxxxxxx)$"> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>> -- >>>>>>> Michael S Collins >>>>>>> Twitter: @mercutioviz >>>>>>> http://www.FreeSWITCH.org >>>>>>> http://www.ClueCon.com >>>>>>> http://www.OSTAG.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130523/9058b7c1/attachment-0001.html From netcentrica at gmail.com Thu May 23 15:47:54 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Thu, 23 May 2013 13:47:54 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: Finally solved this problem by: 1. outbound-codec-prefs to PCMU at 20i,PCMA at 20i 2. setting inbound-codec-negotiation = scrooge Spent 4 hours on fixing that, maybe my solution will help someone with same problem in the future -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/e4646237/attachment.html From sertys at gmail.com Thu May 23 16:01:54 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Thu, 23 May 2013 14:01:54 +0200 Subject: [Freeswitch-users] SDP manipulation and sofia_glue.c (developer help) Message-ID: I am having difficulties forcefully stripping the a:crypto lines from the remote sdp string. At first i was doing from the diaplan, but this breaks the proxy media easily, because it doesn't patch the SDP later for glueing. I commented the following lines where the patcher is checking if the sdp string has been set before : void sofia_glue_tech_patch_sdp(private_object_t *tech_pvt) { switch_size_t len; char *p, *q, *pe, *qe; int has_video = 0, has_audio = 0, has_ip = 0; char port_buf[25] = ""; char vport_buf[25] = ""; char *new_sdp; int bad = 0; */** * if (zstr(tech_pvt->local_sdp_str)) {* * return;* * }* **/* And now i got the correct sdp sent to the b-leg(patched for proxy media and nat), but no audio is going either way. Would there be a more clever way to "rip" some lines from the SDP and not break the NAT/Proxy media processing afterwards? Help is much appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/51079d10/attachment.html From trever.adams at gmail.com Thu May 23 16:28:02 2013 From: trever.adams at gmail.com (Trever L. Adams) Date: Thu, 23 May 2013 06:28:02 -0600 Subject: [Freeswitch-users] inband DTMF bleeding through In-Reply-To: References: <514C8CB9.2010405@gmail.com> Message-ID: <519E0B52.6040700@gmail.com> On 05/22/2013 05:51 PM, Richard Zheng wrote: > Any resolution on this one? I have the same issue. DTMF coming from > a cell phone, say 123, is sometimes repeated, e.g. 11233 when > forwarded to an outbound call. It is not consistent though. > I have not yet tried what was suggested. I believe it works, but I just haven't got around to trying it yet as I have a lot on my plate and FreeSWITCH isn't a top priority yet. freetdm_disable_dtmf=true freetdm_pre_buffer_size=N (size in bytes of a SLIN pre buffer so it can cut detected dtmf fragments out) I do not know if there is any documentation for those yet or not. I hope it helps. Trever -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 263 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/93a1bcdb/attachment.bin From gmangudai at gmail.com Thu May 23 04:42:27 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Thu, 23 May 2013 08:42:27 +0800 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference In-Reply-To: References: Message-ID: sorry for the late response, normally the caller C will make an inbound call through a certain number or preferably a prefix+caller A or B's number (e.g. **+1001), i also would like to know what should i do if caller A or B would dial a prefix+caller C's number and pull him into a three-way call. many thanks! 2013/5/21 Brian Foster > mod_spy or have your client manage a 3 way call. Please explain your > situation a little more we might be able to give you a better answer. Is > caller C inbound or outbound to the call? > On May 21, 2013 8:55 AM, "Vincent Xia" wrote: > >> Dear Support, >> >> my question for using freeswitch is: >> >> how to turn an ongoing call in to a conference so i can have the third >> party in without breaking up the call (no hangup, dial stuff)? >> something like call intrusion?A and B are already talking on the phone, >> then C wants to talk to both A and B. >> >> any response will be greatly appriciated. >> >> Jacob >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/652bf6ea/attachment.html From gmangudai at gmail.com Thu May 23 06:12:56 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Thu, 23 May 2013 10:12:56 +0800 Subject: [Freeswitch-users] why it hangs up after originate Message-ID: when im executing console commands like "originate user/1005 @eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i click answer and find out it hangs up immediately, the freeswitch log says "[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005 at IPaddr:5088 has executed the last dialplan instruction, hanging up." does that mean i need to add something the to the dialplan? is there a simple solution by extending the originate command like "originate user/1005 @eavesdrop(uuid) &wait_for_hangup"? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/4db0068f/attachment.html From cordes.matthew at gmail.com Thu May 23 10:00:17 2013 From: cordes.matthew at gmail.com (Matthew Cordes) Date: Thu, 23 May 2013 02:00:17 -0400 Subject: [Freeswitch-users] Conference questions Message-ID: Hi, I'm new to FreeSwitch and starting to experiment with conferences. I've browsed the wiki and I'm half way through the ebook and I'm running into a few questions I'm hoping I might be able to get some help with. 1. How do I translate conference cli commands (e.g. "conference xxx mute yyy") into a dialplan xml actions? 2. The default behavior of the conference is to disable hold music and allow the participants to speak when there are two or more callers. How might I keep the conference on hold until a particular caller joins? Additionally when this special user leaves I'd like to place the conference back on hold. 3. Where might I find info regarding loading configuration information (particularly the user directory) from an external source (probably a database and probably via mod_python)? 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? Thanks, -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/cdfb2af0/attachment.html From mike at jerris.com Thu May 23 16:28:35 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 08:28:35 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> Message-ID: <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> Did you ever post a new log after you changed codec negotiation settings? On May 23, 2013, at 1:27 AM, Hermouet Erwan wrote: > Up please. It very urgent > > Tks > > > Brian Foster a ?crit : > No, you will not need to use that. > > On May 21, 2013 8:00 AM, wrote: > I must use ? if yes where > and how ? > > tks advance for your help > > > Le 2013-05-20 23:11, Brian Foster a ?crit : > > 2013-05-17 09:39:54.214928 [WARNING] mod_sofia.c:1363 Pass 2833 mode > > may not work on a transcoded call. > > > > You shouldnt be transcoding if you can help it. Now, Im not sure if > > that is an empty threat but you should enable late codec negotiation. > > Information can be found here: > > http://wiki.freeswitch.org/wiki/Codec_negotiation [40] > > > > -BDF > > On May 17, 2013 3:48 AM, wrote: > > > >> im so stupid :) > >> tks > >> > >> http://pastebin.freeswitch.org/20933 [1] > >> > >> called num is 022206... and when i try to use dtmf touch 5 its not > >> works. > >> > >> tks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/b5d1c50a/attachment-0001.html From mike at jerris.com Thu May 23 16:30:58 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 08:30:58 -0400 Subject: [Freeswitch-users] Max. Number of 8 span PRI cards In-Reply-To: References: Message-ID: <4FA13375-1187-461E-AAFB-8B0951CFADAB@jerris.com> Talk to sangoma, they have tested at > 3 cards I know, but I don't know system specifics. On May 23, 2013, at 7:27 AM, Ashish gautam wrote: > Hi, > > I want to ask that is it possible to use 3 eight span PRI cards on a single FS server? I mean is there any limit up to which the system performs optimally on this? > > Please throw some light on this. > > Thanks > > -- > Ashish Gautam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/e975e6c2/attachment.html From ehermouet at bluetel.fr Thu May 23 16:44:22 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 14:44:22 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> Message-ID: Yes http://pastebin.freeswitch.org/20947 Le 2013-05-23 14:28, Michael Jerris a ?crit?: > Did you ever post a new log after you changed codec negotiation > settings? > > On May 23, 2013, at 1:27 AM, Hermouet Erwan [7]> > wrote: > >> Up please. It very urgent >> >> Tks >> >> Brian Foster a ?crit : >> >>> No, you will not need to use that. >>> On May 21, 2013 8:00 AM, wrote: >>> >>>> I must use ? if yes >>>> where >>>> and how ? >>>> >>>> tks advance for your help >>>> >>>> Le 2013-05-20 23:11, Brian Foster a ?crit : >>>> > 2013-05-17 09 [1]:39:54.214928 [WARNING] mod_sofia.c:1363 >>>> Pass 2833 mode >>>> > may not work on a transcoded call. >>>> > >>>> > You shouldnt be transcoding if you can help it. Now, Im not >>>> sure if >>>> > that is an empty threat but you should enable late codec >>>> negotiation. >>>> > Information can be found here: >>>> > http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >>>> > >>>> > -BDF >>>> > On May 17, 2013 3:48 AM, >>>> wrote: >>>> > >>>> >> im so stupid :) >>>> >> tks >>>> >> >>>> >> http://pastebin.freeswitch.org/20933 [4] [1] >>>> >> >>>> >> called num is 022206... and when i try to use dtmf touch 5 >>>> its not >>>> >> works. >>>> >> >>>> >> tks > > > > Links: > ------ > [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 > [2] http://wiki.freeswitch.org/wiki/Codec_negotiation > [3] mailto:ehermouet at bluetel.fr > [4] http://pastebin.freeswitch.org/20933 > [5] mailto:ehermouet at bluetel.fr > [6] mailto:bdfoster at davri.com > [7] mailto:ehermouet at bluetel.fr From mike at jerris.com Thu May 23 16:51:09 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 08:51:09 -0400 Subject: [Freeswitch-users] Conference questions In-Reply-To: References: Message-ID: <7691BDFF-1350-4B32-B990-A513228CE006@jerris.com> On May 23, 2013, at 2:00 AM, Matthew Cordes wrote: > Hi, > > I'm new to FreeSwitch and starting to experiment with conferences. I've browsed the wiki and I'm half way through the ebook and I'm running into a few questions I'm hoping I might be able to get some help with. > > 1. How do I translate conference cli commands (e.g. "conference xxx mute yyy") into a dialplan xml actions? You don't really. Why exactly would you want to do this? > > 2. The default behavior of the conference is to disable hold music and allow the participants to speak when there are two or more callers. How might I keep the conference on hold until a particular caller joins? Additionally when this special user leaves I'd like to place the conference back on hold. https://wiki.freeswitch.org/wiki/Mod_conference conference flags "wait-mod" > > 3. Where might I find info regarding loading configuration information (particularly the user directory) from an external source (probably a database and probably via mod_python)? my recommendation : https://wiki.freeswitch.org/wiki/Mod_xml_curl > > 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? If your doing dialplan via mod_xml_curl you can do all that validation in your dialplan lookup. > > Thanks, > -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/ad866f14/attachment.html From mike at jerris.com Thu May 23 16:58:42 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 08:58:42 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> Message-ID: <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> this log does not seem to have a complete call let alone any attempt at dtmf. I don't see anything wrong from this log but as I said, its incomplete. If you pcap the traffic, do you see 2833 dtmf flowing ? Mike On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: > Yes > > http://pastebin.freeswitch.org/20947 > > Le 2013-05-23 14:28, Michael Jerris a ?crit : >> Did you ever post a new log after you changed codec negotiation >> settings? >> >> On May 23, 2013, at 1:27 AM, Hermouet Erwan > [7]> >> wrote: >> >>> Up please. It very urgent >>> >>> Tks >>> >>> Brian Foster a ?crit : >>> >>>> No, you will not need to use that. >>>> On May 21, 2013 8:00 AM, wrote: >>>> >>>>> I must use ? if yes >>>>> where >>>>> and how ? >>>>> >>>>> tks advance for your help >>>>> >>>>> Le 2013-05-20 23:11, Brian Foster a ?crit : >>>>>> 2013-05-17 09 [1]:39:54.214928 [WARNING] mod_sofia.c:1363 >>>>> Pass 2833 mode >>>>>> may not work on a transcoded call. >>>>>> >>>>>> You shouldnt be transcoding if you can help it. Now, Im not >>>>> sure if >>>>>> that is an empty threat but you should enable late codec >>>>> negotiation. >>>>>> Information can be found here: >>>>>> http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >>>>>> >>>>>> -BDF >>>>>> On May 17, 2013 3:48 AM, >>>>> wrote: >>>>>> >>>>>>> im so stupid :) >>>>>>> tks >>>>>>> >>>>>>> http://pastebin.freeswitch.org/20933 [4] [1] >>>>>>> >>>>>>> called num is 022206... and when i try to use dtmf touch 5 >>>>> its not >>>>>>> works. >>>>>>> >>>>>>> tks >> >> >> >> Links: >> ------ >> [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 >> [2] http://wiki.freeswitch.org/wiki/Codec_negotiation >> [3] mailto:ehermouet at bluetel.fr >> [4] http://pastebin.freeswitch.org/20933 >> [5] mailto:ehermouet at bluetel.fr >> [6] mailto:bdfoster at davri.com >> [7] mailto:ehermouet at bluetel.fr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mike at jerris.com Thu May 23 17:00:56 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 09:00:56 -0400 Subject: [Freeswitch-users] why it hangs up after originate In-Reply-To: References: Message-ID: On May 22, 2013, at 10:12 PM, Vincent Xia wrote: > when im executing console commands like "originate user/1005 @eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i click answer and find out it hangs up immediately, the freeswitch log says "[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005 at IPaddr:5088 has executed the last dialplan instruction, hanging up." > > does that mean i need to add something the to the dialplan? is there a simple solution by extending the originate command like "originate user/1005 @eavesdrop(uuid) &wait_for_hangup"? http://wiki.freeswitch.org/wiki/Mod_commands#originate "&" you need & not @ Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/15a056a8/attachment.html From ehermouet at bluetel.fr Thu May 23 17:07:40 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 15:07:40 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> Message-ID: how do you use it without interface ? it's server with only ssh access. tks Le 2013-05-23 14:58, Michael Jerris a ?crit?: > this log does not seem to have a complete call let alone any attempt > at dtmf. I don't see anything wrong from this log but as I said, its > incomplete. If you pcap the traffic, do you see 2833 dtmf flowing ? > > Mike > > On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: > >> Yes >> >> http://pastebin.freeswitch.org/20947 >> >> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>> Did you ever post a new log after you changed codec negotiation >>> settings? >>> >>> On May 23, 2013, at 1:27 AM, Hermouet Erwan >> [7]> >>> wrote: >>> >>>> Up please. It very urgent >>>> >>>> Tks >>>> >>>> Brian Foster a ?crit : >>>> >>>>> No, you will not need to use that. >>>>> On May 21, 2013 8:00 AM, wrote: >>>>> >>>>>> I must use ? if yes >>>>>> where >>>>>> and how ? >>>>>> >>>>>> tks advance for your help >>>>>> >>>>>> Le 2013-05-20 23:11, Brian Foster a ?crit : >>>>>>> 2013-05-17 09 [1]:39:54.214928 [WARNING] mod_sofia.c:1363 >>>>>> Pass 2833 mode >>>>>>> may not work on a transcoded call. >>>>>>> >>>>>>> You shouldnt be transcoding if you can help it. Now, Im not >>>>>> sure if >>>>>>> that is an empty threat but you should enable late codec >>>>>> negotiation. >>>>>>> Information can be found here: >>>>>>> http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >>>>>>> >>>>>>> -BDF >>>>>>> On May 17, 2013 3:48 AM, >>>>>> wrote: >>>>>>> >>>>>>>> im so stupid :) >>>>>>>> tks >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20933 [4] [1] >>>>>>>> >>>>>>>> called num is 022206... and when i try to use dtmf touch 5 >>>>>> its not >>>>>>>> works. >>>>>>>> >>>>>>>> tks >>> >>> >>> >>> Links: >>> ------ >>> [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 >>> [2] http://wiki.freeswitch.org/wiki/Codec_negotiation >>> [3] mailto:ehermouet at bluetel.fr >>> [4] http://pastebin.freeswitch.org/20933 >>> [5] mailto:ehermouet at bluetel.fr >>> [6] mailto:bdfoster at davri.com >>> [7] mailto:ehermouet at bluetel.fr >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mike at jerris.com Thu May 23 17:24:14 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 09:24:14 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> Message-ID: <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> How do you use what? On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: > how do you use it without interface ? it's server with only ssh access. > tks > Le 2013-05-23 14:58, Michael Jerris a ?crit : >> this log does not seem to have a complete call let alone any attempt >> at dtmf. I don't see anything wrong from this log but as I said, its >> incomplete. If you pcap the traffic, do you see 2833 dtmf flowing ? >> >> Mike >> >> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >> >>> Yes >>> >>> http://pastebin.freeswitch.org/20947 >>> >>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>> Did you ever post a new log after you changed codec negotiation >>>> settings? >>>> From ehermouet at bluetel.fr Thu May 23 17:27:11 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 15:27:11 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> Message-ID: <8e544c847cbeb3049bab85cdd50858c3@bluetel.fr> It's not often time but now i see again message error 2013-05-23 15:24:44.452821 [WARNING] mod_sofia.c:1363 Pass 2833 mode may not work on a transcoded call. 2013-05-23 15:24:44.452821 [DEBUG] switch_rtp.c:2534 Correct ip/port confirmed. 2013-05-23 15:24:44.372152 [WARNING] mod_sofia.c:1363 Pass 2833 mode may not work on a transcoded call. here my config file external.xml and internal.xml Tks adavnce for your help Le 2013-05-23 14:58, Michael Jerris a ?crit?: > this log does not seem to have a complete call let alone any attempt > at dtmf. I don't see anything wrong from this log but as I said, its > incomplete. If you pcap the traffic, do you see 2833 dtmf flowing ? > > Mike > > On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: > >> Yes >> >> http://pastebin.freeswitch.org/20947 >> >> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>> Did you ever post a new log after you changed codec negotiation >>> settings? >>> >>> On May 23, 2013, at 1:27 AM, Hermouet Erwan >> [7]> >>> wrote: >>> >>>> Up please. It very urgent >>>> >>>> Tks >>>> >>>> Brian Foster a ?crit : >>>> >>>>> No, you will not need to use that. >>>>> On May 21, 2013 8:00 AM, wrote: >>>>> >>>>>> I must use ? if yes >>>>>> where >>>>>> and how ? >>>>>> >>>>>> tks advance for your help >>>>>> >>>>>> Le 2013-05-20 23:11, Brian Foster a ?crit : >>>>>>> 2013-05-17 09 [1]:39:54.214928 [WARNING] mod_sofia.c:1363 >>>>>> Pass 2833 mode >>>>>>> may not work on a transcoded call. >>>>>>> >>>>>>> You shouldnt be transcoding if you can help it. Now, Im not >>>>>> sure if >>>>>>> that is an empty threat but you should enable late codec >>>>>> negotiation. >>>>>>> Information can be found here: >>>>>>> http://wiki.freeswitch.org/wiki/Codec_negotiation [2] [40] >>>>>>> >>>>>>> -BDF >>>>>>> On May 17, 2013 3:48 AM, >>>>>> wrote: >>>>>>> >>>>>>>> im so stupid :) >>>>>>>> tks >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20933 [4] [1] >>>>>>>> >>>>>>>> called num is 022206... and when i try to use dtmf touch 5 >>>>>> its not >>>>>>>> works. >>>>>>>> >>>>>>>> tks >>> >>> >>> >>> Links: >>> ------ >>> [1] http://webmail.gandi.net/tel:2013-05-17%C2%A009 >>> [2] http://wiki.freeswitch.org/wiki/Codec_negotiation >>> [3] mailto:ehermouet at bluetel.fr >>> [4] http://pastebin.freeswitch.org/20933 >>> [5] mailto:ehermouet at bluetel.fr >>> [6] mailto:bdfoster at davri.com >>> [7] mailto:ehermouet at bluetel.fr >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From cordes.matthew at gmail.com Thu May 23 17:57:50 2013 From: cordes.matthew at gmail.com (Matthew Cordes) Date: Thu, 23 May 2013 09:57:50 -0400 Subject: [Freeswitch-users] Conference questions Message-ID: Hi Mike, Thanks for the quick response. Here are a few more questions / comments below: > 1. How do I translate conference cli commands (e.g. "conference xxx mute > yyy") into a dialplan xml actions? > > You don't really. Why exactly would you want to do this? > In some situations I'd like to mute all users other than the moderator from when they join for the duration of the conference. I thought the right way to do this would be via an action in my dialplan as new callers join my conference. What would you recommend? Thanks, Mod_xml_curl looks straightforward enough. 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? If your doing dialplan via mod_xml_curl you can do all that validation in > your dialplan lookup. > Can you point me to a resource regarding how I would do this validation? I know how to define users and groups. It's not clear to me how to test for membership in a group or how that would be written (although I'm assuming it would be a condition in the dialplan, correct?). -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/cf07b2d9/attachment.html From ehermouet at bluetel.fr Thu May 23 18:05:56 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 16:05:56 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> Message-ID: <9e397857971309b1cf47340345721e94@bluetel.fr> pcap. i send you the xml file and log in my previous email... because i see problem sometime... i'm sure i have error on my xml file. can you check it. ? tks Le 2013-05-23 15:24, Michael Jerris a ?crit?: > How do you use what? > > On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: > >> how do you use it without interface ? it's server with only ssh >> access. >> tks >> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>> this log does not seem to have a complete call let alone any >>> attempt >>> at dtmf. I don't see anything wrong from this log but as I said, >>> its >>> incomplete. If you pcap the traffic, do you see 2833 dtmf flowing >>> ? >>> >>> Mike >>> >>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>> >>>> Yes >>>> >>>> http://pastebin.freeswitch.org/20947 >>>> >>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>> Did you ever post a new log after you changed codec negotiation >>>>> settings? >>>>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mike at hendrienet.com Thu May 23 18:18:29 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Thu, 23 May 2013 09:18:29 -0500 Subject: [Freeswitch-users] Cannot ring extension from DID In-Reply-To: References: <519A9A81.7020609@ppmt.org> <519DF9B9.5010506@ppmt.org> Message-ID: Looking at the log: http://pastebin.freeswitch.org/20964 Am I seeing the call transferring to 1000 at default and not GothamCity.xom? Dialplan: sofia/external/5555555555 at 66.66.66.66 Regex (PASS) [GothamCity.xom 1001] destination_number(2222222222) =~ /^1?(2222222222)$/ break=on-false Dialplan: sofia/external/5555555555 at 66.66.66.66 Action set(domain_name=GothamCity.xom) Dialplan: sofia/external/5555555555 at 66.66.66.66 Action transfer(1000 XML default) The Dialplan should have it go to 1000 at GithamCity.xom Thanks! On Thu, May 23, 2013 at 6:45 AM, Mike Hendrie wrote: > I made a change to 1000 this morning and I can now dial each others > extensions. > > When I call the number I still go directly to the voicemail when extension > 1000 should ring. > > > freeswitch at internal> sofia status profile internal reg > > > Registrations: > > > ================================================================================================= > > Call-ID: 519830183003 at 10.2.1.209 > > User: 1001 at gothamcity.xom > > Contact: "user" ;transport=udp;fs_nat=yes;fs_path=sip%3A1001%4010.2.1.209%3A45048%3Btransport%3Dudp> > > Agent: Sipdroid/3.0 beta/SCH-I605 > > Status: Registered(UDP-NAT)(unknown) EXP(2013-05-22 23:44:29) > EXPSECS(3501) > > Host: GothamCity-00 > > IP: 10.2.1.209 > > Port: 45048 > > Auth-User: 1001 > > Auth-Realm: gothamcity.xom > > MWI-Account: 1001 at gothamcity.xom > > > Call-ID: 7eb5078e-316ad43b-c274631c at 10.2.1.50 > > User: 1000 at gothamcity.xom > > Contact: "user" ;transport=tcp;fs_nat=yes;fs_path=sip%3A1000%4010.2.1.50%3A64461%3Btransport%3Dtcp> > > Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 > > Status: Registered(TCP-NAT)(unknown) EXP(2013-05-22 22:48:44) > EXPSECS(156) > > Host: GothamCity-00 > > IP: 10.2.1.50 > > Port: 64461 > > Auth-User: 1000 > > Auth-Realm: gothamcity.xom > > MWI-Account: 1000 at gothamcity.xom > > > Total items returned: 2 > > > ================================================================================================= > > > Mike Hendrie > T: 847.366.5881 > E: mike at hendrienet.com > On May 23, 2013 6:16 AM, "Philippe Le Toquin" wrote: > >> so your extension are not registered >> >> On my FS I can see my own extension. >> >> Now this is where I am going to learn something because I have no idea >> why you can call from one extension to another if they are not registered. >> >> On 13-05-22 10:48 PM, Mike Hendrie wrote: >> >> Sorry it took so long. >> >> I updated the log file and your requested details. >> >> http://pastebin.freeswitch.org/20963 >> >> Thank you >> >> >> On Wed, May 22, 2013 at 10:38 AM, Michael Collins wrote: >> >>> Mike, >>> >>> You might be confusing "profiles" with "domains". A profile is a "SIP >>> profile" and represent a SIP user agent where FreeSWITCH's SIP stack >>> ("Sofia") listens and responds. There is no explicit relationship between >>> domains and profiles. Profiles simply define where SIP messages come and go >>> on the system. >>> >>> Domains are a logical construct in the server. You can have multiple >>> domains and thus multiple users with the same ID, i.e. "1001 at domain1" >>> and "1001 at domain2". >>> >>> Go to your fs_cli and capture the output of these commands: >>> sofia status >>> sofia status profile internal reg >>> >>> Drop those in a pastebin a link back here. >>> -MC >>> >>> >>> On Wed, May 22, 2013 at 7:14 AM, Mike Hendrie wrote: >>> >>>> I will look into the registration of the phones as I see I am missing >>>> something. >>>> >>>> freeswitch at internal> sofia status profile GothamCity.xom reg >>>> >>>> Registrations: >>>> >>>> ================================================================================================= >>>> Total items returned: 0 >>>> >>>> ================================================================================================= >>>> >>>> >>>> >>>> On Wed, May 22, 2013 at 8:15 AM, Mike Hendrie wrote: >>>> >>>>> I do. I can also ring extension 1000. >>>>> On May 22, 2013 7:52 AM, "Philippe Le Toquin" >>>>> wrote: >>>>> >>>>>> This is why it goes to voicemail I guess >>>>>> >>>>>> 1. 2013-05-21 22:28:54.107400 [DEBUG] switch_channel.c:1099sofia/external/ >>>>>> 5555555555 at 66.66.66.66 EXPORTING[export_vars] [dialed_extension]=[ >>>>>> 1001] to event >>>>>> 2. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c: >>>>>> 2044 Parsing global variables >>>>>> 3. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608Parsing variable >>>>>> [sip_invite_domain]=[GothamCity.xom] >>>>>> 4. 2013-05-21 22:28:54.107400 [DEBUG] switch_event.c:1608Parsing variable >>>>>> [presence_id]=[1001 at GothamCity.xom] >>>>>> 5. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c: >>>>>> 2639 Cannot create outgoing channel of type [error] cause: [ >>>>>> USER_NOT_REGISTERED] >>>>>> 6. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c: >>>>>> 3605 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >>>>>> 7. 2013-05-21 22:28:54.107400 [NOTICE] switch_ivr_originate.c: >>>>>> 2639 Cannot create outgoing channel of type [user] cause: [ >>>>>> USER_NOT_REGISTERED] >>>>>> 8. 2013-05-21 22:28:54.107400 [DEBUG] switch_ivr_originate.c: >>>>>> 3605 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >>>>>> 9. 2013-05-21 22:28:54.107400 [INFO] mod_dptools.c:3106Originate Failed. Cause: USER_NOT_REGISTERED >>>>>> 10. EXECUTE sofia/external/5555555555 at 66.66.66.66 answer() >>>>>> 11. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>>>> [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] >>>>>> 12. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:2663Audio Codec Compare >>>>>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>>>> 13. 2013-05-21 22:28:54.107400 [DEBUG] switch_core_media.c:1772Set Codec sofia/external/ >>>>>> 5555555555 at 66.66.66.66 P >>>>>> >>>>>> Do you have a dialtone on that 1001 extension? >>>>>> >>>>>> >>>>>> On 22 May 2013 07:55, Mike Hendrie wrote: >>>>>> >>>>>>> Thank you. >>>>>>> >>>>>>> Here is the log URL. >>>>>>> http://pastebin.freeswitch.org/20960 >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, May 22, 2013 at 1:39 AM, Michael Collins < >>>>>>> msc at freeswitch.org> wrote: >>>>>>> >>>>>>>> Post a FreeSWITCH debug log of the incoming call. Use >>>>>>>> pastebin.freeswitch.org and select "FreeSWITCH Log" as the syntax >>>>>>>> highlighting. Paste the URL in this email thread and we'll take a look. >>>>>>>> -MC >>>>>>>> >>>>>>>> >>>>>>>> On Tue, May 21, 2013 at 9:35 PM, Mike Hendrie wrote: >>>>>>>> >>>>>>>>> Correction: >>>>>>>>> I had a second dialplan in the public folder that was causing >>>>>>>>> confusion. Below is the dialplan I am using. >>>>>>>>> If I change the extension in the dialplan from 1000 to 1001 I >>>>>>>>> get the appropriate voice mail extension, however, the phones never ring. >>>>>>>>> >>>>>>>>> I have the fs configured as a multi-tenant solution. >>>>>>>>> >>>>>>>>> Could the dialplan be using the default extensions (1000 and >>>>>>>>> 1001) under /conf/directory/default and not reference the >>>>>>>>> /conf/directory/GothamCity.xom domain? That would explain why I get to the >>>>>>>>> voicemail for the correct extension when the phone never rings. >>>>>>>>> >>>>>>>>> Thanks >>>>>>>>> >>>>>>>>> ===== >>>>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> data="domain_name=GothamCity.xom"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> ===== >>>>>>>>> /conf/directory/GothamCity.xom.xml >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@ >>>>>>>>> ${dialed_domain}}${sofia_contact(*/${dialed_user}@ >>>>>>>>> ${dialed_domain})}"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> value="$${default_provider}"/> >>>>>>>>> >>>>>>>> value="$${default_areacode}"/> >>>>>>>>> >>>>>>>> value="operator"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> data="GothamCity.xom/*.xml"/> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> ===== >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, May 21, 2013 at 9:27 PM, Mike Hendrie >>>>>>>> > wrote: >>>>>>>>> >>>>>>>>>> Thank you for your assistance. I made the suggested modification >>>>>>>>>> below, however, when calling the number it goes directly to voicemail. >>>>>>>>>> >>>>>>>>>> /usr/local/freeswitch/conf/dialplan/public/GothamCity.xom.xml >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> expression="^1?(262xxxxxxx)$"> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>> -- >>>>>>>> Michael S Collins >>>>>>>> Twitter: @mercutioviz >>>>>>>> http://www.FreeSWITCH.org >>>>>>>> http://www.ClueCon.com >>>>>>>> http://www.OSTAG.org >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130523/f5e8d01d/attachment-0001.html From krice at freeswitch.org Thu May 23 18:49:06 2013 From: krice at freeswitch.org (Ken Rice) Date: Thu, 23 May 2013 09:49:06 -0500 Subject: [Freeswitch-users] FreeSwitch Proxy + RTPProxy Media server In-Reply-To: <1369300391544-7590996.post@n2.nabble.com> Message-ID: This will absolutely work. You just have to set up the dialplans properly for the calls to be routed... On 5/23/13 4:13 AM, "adahary" wrote: > Ken, > > It does not work for me. please check the log below. > ENDPOINT.1 = 1001 > ENDPOINT.2 = 1000 > > 2013-05-23 13:10:01.984411 [INFO] mod_dialplan_xml.c:558 Processing 1001 > <1001>->1000 in context confitele_endpoints > 2013-05-23 13:10:02.004372 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/42341cdc-9982-4845-88be-0576fbd60ea7.tmp.xml > Dialplan: sofia/endpoint-nat/1001 at fs-a.com parsing > [confitele_endpoints->call_inbound] continue=false > Dialplan: sofia/endpoint-nat/1001 at fs-a.com Regex (PASS) [call_inbound] > destination_number(1000) =~ /^(\d+)$/ break=on-false > Dialplan: sofia/endpoint-nat/1001 at fs-a.com Action set(bypass_media=true) > INLINE > EXECUTE sofia/endpoint-nat/1001 at fs-a.com set(bypass_media=true) > 2013-05-23 13:10:02.004372 [DEBUG] mod_dptools.c:1373 > sofia/endpoint-nat/1001 at fs-a.com SET [bypass_media]=[true] > Dialplan: sofia/endpoint-nat/1001 at fs-a.com Action > bridge(${destination_number}@fs-b.com) > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:167 > (sofia/endpoint-nat/1001 at fs-a.com) State Change CS_ROUTING -> CS_EXECUTE > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_session.c:1340 Send signal > sofia/endpoint-nat/1001 at fs-a.com [BREAK] > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:470 > (sofia/endpoint-nat/1001 at fs-a.com) State ROUTING going to sleep > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:415 > (sofia/endpoint-nat/1001 at fs-a.com) Running State Change CS_EXECUTE > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:477 > (sofia/endpoint-nat/1001 at fs-a.com) State EXECUTE > 2013-05-23 13:10:02.004372 [DEBUG] mod_sofia.c:230 > sofia/endpoint-nat/1001 at fs-a.com SOFIA EXECUTE > 2013-05-23 13:10:02.004372 [DEBUG] switch_core_state_machine.c:209 > sofia/endpoint-nat/1001 at fs-a.com Standard EXECUTE > EXECUTE sofia/endpoint-nat/1001 at fs-a.com bridge(1000 at fs-b.com) > 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:2039 Parsing > global variables > 2013-05-23 13:10:02.004372 [ERR] switch_core_session.c:496 Could not locate > channel type 1000 at fs-b.com > 2013-05-23 13:10:02.004372 [NOTICE] switch_ivr_originate.c:2649 Cannot > create outgoing channel of type [1000 at fs-b.com] cause: > [CHAN_NOT_IMPLEMENTED] > 2013-05-23 13:10:02.004372 [DEBUG] switch_ivr_originate.c:3615 Originate > Resulted in Error Cause: 66 [CHAN_NOT_IMPLEMENTED] > 2013-05-23 13:10:02.004372 [INFO] mod_dptools.c:3098 Originate Failed. > Cause: CHAN_NOT_IMPLEMENTED > 2013-05-23 13:10:02.004372 [NOTICE] mod_dptools.c:3218 Hangup > sofia/endpoint-nat/1001 at fs-a.com [CS_EXECUTE] [CHAN_NOT_IMPLEMENTED] > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FreeSwitch-Proxy-RTPProxy-Media- > server-tp7590972p7590996.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From mike at jerris.com Thu May 23 18:58:48 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 10:58:48 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <9e397857971309b1cf47340345721e94@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> Message-ID: you can use tcpdump to get a pcap. I didn't see anything wrong in review of what you have posted so far. On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: > pcap. > > i send you the xml file and log in my previous email... because i see > problem sometime... i'm sure i have error on my xml file. can you check > it. ? > > tks > > Le 2013-05-23 15:24, Michael Jerris a ?crit : >> How do you use what? >> >> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >> >>> how do you use it without interface ? it's server with only ssh >>> access. >>> tks >>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>> this log does not seem to have a complete call let alone any >>>> attempt >>>> at dtmf. I don't see anything wrong from this log but as I said, >>>> its >>>> incomplete. If you pcap the traffic, do you see 2833 dtmf flowing >>>> ? >>>> >>>> Mike >>>> >>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>> >>>>> Yes >>>>> >>>>> http://pastebin.freeswitch.org/20947 >>>>> >>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>> Did you ever post a new log after you changed codec negotiation >>>>>> settings? >>>>>> >> >> From mike at jerris.com Thu May 23 19:01:11 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 11:01:11 -0400 Subject: [Freeswitch-users] Conference questions In-Reply-To: References: Message-ID: On May 23, 2013, at 9:57 AM, Matthew Cordes wrote: > Hi Mike, > > Thanks for the quick response. Here are a few more questions / comments below: >> 1. How do I translate conference cli commands (e.g. "conference xxx mute yyy") into a dialplan xml actions? > You don't really. Why exactly would you want to do this? > > In some situations I'd like to mute all users other than the moderator from when they join for the duration of the conference. I thought the right way to do this would be via an action in my dialplan as new callers join my conference. What would you recommend? Look at conference flags, there is one so they enter muted. > Thanks, Mod_xml_curl looks straightforward enough. > >> 4. I'd like to have some private conferences that only certain callers can join. I'll know the callers' ids before hand. I'm assuming a reasonable way to handle is with user groups. How would I test if a user is a member of a particular group? > If your doing dialplan via mod_xml_curl you can do all that validation in your dialplan lookup. > > Can you point me to a resource regarding how I would do this validation? I know how to define users and groups. It's not clear to me how to test for membership in a group or how that would be written (although I'm assuming it would be a condition in the dialplan, correct?). This has nothing to do with dialplan, it has to do with however you store data about these "groups". This is completely arbitrary outside of freeswitch. You could store this information in a database and look it up from your script managing dialplan for example. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/5fa92fae/attachment.html From ehermouet at bluetel.fr Thu May 23 19:06:15 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 17:06:15 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> Message-ID: Ok i will test. but Did you read that ? It's not often time but now i see again message error 2013-05-23 15:24:44.452821 [WARNING] mod_sofia.c:1363 Pass 2833 mode may not work on a transcoded call. 2013-05-23 15:24:44.452821 [DEBUG] switch_rtp.c:2534 Correct ip/port confirmed. 2013-05-23 15:24:44.372152 [WARNING] mod_sofia.c:1363 Pass 2833 mode may not work on a transcoded call. here my config file external.xml and internal.xml Tks adavnce for your help Le 2013-05-23 16:58, Michael Jerris a ?crit?: > you can use tcpdump to get a pcap. I didn't see anything wrong in > review of what you have posted so far. > > On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: > >> pcap. >> >> i send you the xml file and log in my previous email... because i >> see >> problem sometime... i'm sure i have error on my xml file. can you >> check >> it. ? >> >> tks >> >> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>> How do you use what? >>> >>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>> >>>> how do you use it without interface ? it's server with only ssh >>>> access. >>>> tks >>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>> this log does not seem to have a complete call let alone any >>>>> attempt >>>>> at dtmf. I don't see anything wrong from this log but as I said, >>>>> its >>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>> flowing >>>>> ? >>>>> >>>>> Mike >>>>> >>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> Yes >>>>>> >>>>>> http://pastebin.freeswitch.org/20947 >>>>>> >>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>> Did you ever post a new log after you changed codec negotiation >>>>>>> settings? >>>>>>> >>> >>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ehermouet at bluetel.fr Thu May 23 19:37:31 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Thu, 23 May 2013 17:37:31 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> Message-ID: <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> you can found here my tcpdump file http://bluetelconnect.fr/tcpdump.log tks advance Michael Le 2013-05-23 16:58, Michael Jerris a ?crit?: > you can use tcpdump to get a pcap. I didn't see anything wrong in > review of what you have posted so far. > > On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: > >> pcap. >> >> i send you the xml file and log in my previous email... because i >> see >> problem sometime... i'm sure i have error on my xml file. can you >> check >> it. ? >> >> tks >> >> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>> How do you use what? >>> >>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>> >>>> how do you use it without interface ? it's server with only ssh >>>> access. >>>> tks >>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>> this log does not seem to have a complete call let alone any >>>>> attempt >>>>> at dtmf. I don't see anything wrong from this log but as I said, >>>>> its >>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>> flowing >>>>> ? >>>>> >>>>> Mike >>>>> >>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> Yes >>>>>> >>>>>> http://pastebin.freeswitch.org/20947 >>>>>> >>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>> Did you ever post a new log after you changed codec negotiation >>>>>>> settings? >>>>>>> >>> >>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mike at jerris.com Thu May 23 20:06:01 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 23 May 2013 12:06:01 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> Message-ID: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Have you looked at it to see if it is sending the dtmf? On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > you can found here my tcpdump file > > http://bluetelconnect.fr/tcpdump.log > > tks advance Michael > > > Le 2013-05-23 16:58, Michael Jerris a ?crit : >> you can use tcpdump to get a pcap. I didn't see anything wrong in >> review of what you have posted so far. >> >> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >> >>> pcap. >>> >>> i send you the xml file and log in my previous email... because i >>> see >>> problem sometime... i'm sure i have error on my xml file. can you >>> check >>> it. ? >>> >>> tks >>> >>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>> How do you use what? >>>> >>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>> >>>>> how do you use it without interface ? it's server with only ssh >>>>> access. >>>>> tks >>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>> this log does not seem to have a complete call let alone any >>>>>> attempt >>>>>> at dtmf. I don't see anything wrong from this log but as I said, >>>>>> its >>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>> flowing >>>>>> ? >>>>>> >>>>>> Mike >>>>>> >>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>> >>>>>>> Yes >>>>>>> >>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>> >>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>> Did you ever post a new log after you changed codec negotiation >>>>>>>> settings? From anthony.minessale at gmail.com Thu May 23 20:52:04 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 May 2013 11:52:04 -0500 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference In-Reply-To: References: Message-ID: As explained, save the uuid of either a or b and have c execute "three_way" on that uuid. On Wed, May 22, 2013 at 7:42 PM, Vincent Xia wrote: > sorry for the late response, normally the caller C will make an inbound > call through a certain number or preferably a prefix+caller A or B's number > (e.g. **+1001), i also would like to know what should i do if caller A or B > would dial a prefix+caller C's number and pull him into a three-way call. > > many thanks! > > > > 2013/5/21 Brian Foster > >> mod_spy or have your client manage a 3 way call. Please explain your >> situation a little more we might be able to give you a better answer. Is >> caller C inbound or outbound to the call? >> On May 21, 2013 8:55 AM, "Vincent Xia" wrote: >> >>> Dear Support, >>> >>> my question for using freeswitch is: >>> >>> how to turn an ongoing call in to a conference so i can have the third >>> party in without breaking up the call (no hangup, dial stuff)? >>> something like call intrusion?A and B are already talking on the phone, >>> then C wants to talk to both A and B. >>> >>> any response will be greatly appriciated. >>> >>> Jacob >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/f6c2da22/attachment.html From anthony.minessale at gmail.com Thu May 23 21:33:55 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 May 2013 12:33:55 -0500 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: Sounds like the linksys bug, do you have linksys or sipuras on the other end. The setting on the ptime there has a bug and you have to manually set it to 20. http://wiki.freeswitch.org/wiki/Interop_List#Linksys_Products Our ability to tolerate that might have an issue if it doesn't auto correct itself but then again why do we always have to marr our code to tolerate 10 year old bugs ;) On Thu, May 23, 2013 at 6:47 AM, Adam Raszynski wrote: > Finally solved this problem by: > 1. outbound-codec-prefs to PCMU at 20i,PCMA at 20i > 2. setting inbound-codec-negotiation = scrooge > > Spent 4 hours on fixing that, maybe my solution will help someone with > same problem in the future > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/f1233b0f/attachment-0001.html From anthony.minessale at gmail.com Thu May 23 22:02:01 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 23 May 2013 13:02:01 -0500 Subject: [Freeswitch-users] Lua creating multiple session's In-Reply-To: References: Message-ID: Yes the constructor is blocking until the session has media or is concluded to have failed to setup (whichever comes first) If you need to do things async you can either use ESL from a remote process. You can also write your app in C so you have more control and ability to do that type of thing. On Thu, May 23, 2013 at 6:36 AM, Jon Sch?pzinsky wrote: > Hello List, > > Im trying to do dial multiple destination through lua, on a single > incoming call. > > I do know that i could do a simple session:execute("bridge", > "dst1,dst2,dst3") but I need to do it in individual session, for > processing I need to do in a later version of the lua script. > > I loop through the destinations that needs to be called, creating a new > session for each destination, and storing that in an array. > Firstly, it seems as freeswitch.Session doesn't reply right away, but > waits for early-media. Thats ok though, but makes dialling mobile devices > a rather long wait. > > The problem is, that when I do the second freeswitch.Session, it seems to > hold up further lua processing, until the last created call is answered. > Is this how its supposed to work. > > Heres my current code: > > local legs = {} > > for key, dev in pairs(dstDevices) do > freeswitch.consoleLog("info", "Lets call " .. dev.username .. " with > tech " .. dev.devicetech .. "\n") > if dev.devicetech == "1" then > freeswitch.consoleLog("info", "calling " .. dev.username .. " on > uasbc\n") > legs[key] = freeswitch.Session("sofia/gateway/uasbc01/" .. > dev.username); > freeswitch.consoleLog("info", "called " .. dev.username .. " on > uasbc\n") > elseif dev.devicetech == "2" then > freeswitch.consoleLog("info", "calling " .. dev.username .. " on > ccsbc, mvno\n") > legs[key] = > freeswitch.Session("{origination_caller_id_name=+xxxxxxxx,origination_calle > r_id_number=+xxxxxxxxx}sofia/gateway/ccsbc01/+xx" .. > string.match(dev.username, "^mvno_(.+)")) > freeswitch.consoleLog("info", "called " .. dev.username .. " on > ccsbc, mvno\n") > end > End > > > freeswitch.consoleLog("info", "Enter loop now\n") > > It doesn't reach the last consoleLog until the last call is answered. > > > > Venlig hilsen/kind regards > > Jon Leren Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/bd9f0515/attachment.html From ehermouet at bluetel.fr Thu May 23 22:14:54 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Thu, 23 May 2013 20:14:54 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Message-ID: <4fd6c0fb-3a11-4793-a58b-83e2a694a598@email.android.com> I look but i don t know what i must search. Do you have example ? Michael Jerris a ?crit?: >Have you looked at it to see if it is sending the dtmf? > >On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > >> you can found here my tcpdump file >> >> http://bluetelconnect.fr/tcpdump.log >> >> tks advance Michael >> >> >> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>> review of what you have posted so far. >>> >>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>> >>>> pcap. >>>> >>>> i send you the xml file and log in my previous email... because i >>>> see >>>> problem sometime... i'm sure i have error on my xml file. can you >>>> check >>>> it. ? >>>> >>>> tks >>>> >>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>> How do you use what? >>>>> >>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> how do you use it without interface ? it's server with only ssh >>>>>> access. >>>>>> tks >>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>> this log does not seem to have a complete call let alone any >>>>>>> attempt >>>>>>> at dtmf. I don't see anything wrong from this log but as I >said, >>>>>>> its >>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>> flowing >>>>>>> ? >>>>>>> >>>>>>> Mike >>>>>>> >>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> Yes >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>> >>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>> Did you ever post a new log after you changed codec >negotiation >>>>>>>>> settings? > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/20129804/attachment.html From j_mj at aol.com Thu May 23 22:38:54 2013 From: j_mj at aol.com (John M) Date: Thu, 23 May 2013 14:38:54 -0400 (EDT) Subject: [Freeswitch-users] Channel variable usage in IVR In-Reply-To: <4fd6c0fb-3a11-4793-a58b-83e2a694a598@email.android.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <4fd6c0fb-3a11-4793-a58b-83e2a694a598@email.android.com> Message-ID: <8D025FBE9958E98-18C4-D119@webmail-m236.sysops.aol.com> Hi list... Is it valid to use a variable inside the digits="" element in an IVR? Eg; The value of ${default_ivr_actionkeys} is: 12345 It works if I put the list of number in the regex manually but when I try using it with the variable it won't work. I need to create a dynamic set of digits options for our IVR feature because different customers set different keypress options and I need to dynamically generate this from a db. Is there a better way? Thanks, Jm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/c4b9c766/attachment.html From msc at freeswitch.org Fri May 24 01:43:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 23 May 2013 14:43:01 -0700 Subject: [Freeswitch-users] Channel variable usage in IVR In-Reply-To: <8D025FBE9958E98-18C4-D119@webmail-m236.sysops.aol.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <4fd6c0fb-3a11-4793-a58b-83e2a694a598@email.android.com> <8D025FBE9958E98-18C4-D119@webmail-m236.sysops.aol.com> Message-ID: AFAIK there is no variable expansion at this point in the IVR processing. If you need dynamic IVR the you're best bet is mod_xml_curl or mod_httapi. The new FS 1.2 book has a great chapter on httapi as well as useful information on serving up dynamic configs via xml curl. -MC On May 23, 2013 11:48 AM, "John M" wrote: > Hi list... > > Is it valid to use a variable inside the digits="" element in an IVR? > > Eg; digits="/^([${default_ivr_actionkeys}])$/" param="transfer $1 XML > features"/> > > The value of ${default_ivr_actionkeys} is: 12345 > > It works if I put the list of number in the regex manually but when I try > using it with the variable it won't work. > > I need to create a dynamic set of digits options for our IVR feature > because different customers set different keypress options and I need to > dynamically generate this from a db. > > Is there a better way? > > Thanks, > Jm > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130523/28c19ebf/attachment-0001.html From gmangudai at gmail.com Fri May 24 04:37:03 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Fri, 24 May 2013 08:37:03 +0800 Subject: [Freeswitch-users] why it hangs up after originate In-Reply-To: References: Message-ID: thank you very much, your suggestion really helps! 2013/5/23 Michael Jerris > > On May 22, 2013, at 10:12 PM, Vincent Xia wrote: > > when im executing console commands like "originate user/1005 > @eavesdrop(uuid)" to have 1005 listen to a call party, 1005 rings then i > click answer and find out it hangs up immediately, the freeswitch log says > "[NOTICE] switch_core_state_machine.c:262 sofia/internal/sip:1005 at IPaddr:5088 > has executed the last dialplan instruction, hanging up." > > does that mean i need to add something the to the dialplan? is there a > simple solution by extending the originate command like "originate > user/1005 @eavesdrop(uuid) &wait_for_hangup"? > > > http://wiki.freeswitch.org/wiki/Mod_commands#originate > > "&" > you need & not @ > > Mike > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/18892388/attachment.html From gmangudai at gmail.com Fri May 24 04:44:35 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Fri, 24 May 2013 08:44:35 +0800 Subject: [Freeswitch-users] how to turn an ongoing call in to a conference In-Reply-To: References: Message-ID: thanks a lot Anthony i've now resolved the problem, i sent that reply according to Brian's response before i saw your one, which is the thing i want, i've asked the same question a long time ago in different approaches as myself had also been trying to find out the solution, i got help from many other people and finally get the solution from you, thanks again! 2013/5/24 Anthony Minessale > As explained, save the uuid of either a or b and have c execute > "three_way" on that uuid. > > > > On Wed, May 22, 2013 at 7:42 PM, Vincent Xia wrote: > >> sorry for the late response, normally the caller C will make an inbound >> call through a certain number or preferably a prefix+caller A or B's number >> (e.g. **+1001), i also would like to know what should i do if caller A or B >> would dial a prefix+caller C's number and pull him into a three-way call. >> >> many thanks! >> >> >> >> 2013/5/21 Brian Foster >> >>> mod_spy or have your client manage a 3 way call. Please explain your >>> situation a little more we might be able to give you a better answer. Is >>> caller C inbound or outbound to the call? >>> On May 21, 2013 8:55 AM, "Vincent Xia" wrote: >>> >>>> Dear Support, >>>> >>>> my question for using freeswitch is: >>>> >>>> how to turn an ongoing call in to a conference so i can have the third >>>> party in without breaking up the call (no hangup, dial stuff)? >>>> something like call intrusion?A and B are already talking on the phone, >>>> then C wants to talk to both A and B. >>>> >>>> any response will be greatly appriciated. >>>> >>>> Jacob >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/d42405ed/attachment.html From jalsot at gmail.com Fri May 24 09:30:32 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Fri, 24 May 2013 07:30:32 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: Hello, Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on host and Ubuntu 10.04 LTS in VE. Do you know maybe how to allow realtime priority in the VE for FS? Running FS with -rp does not set the scheduler. strace says, sched_setscheduler operation permitted, so SCHED_FF is not set. Tried to run as root and/or use ulimit -r option, but cannot run FS with tuned priorities.We guess, some thing missing in the host/VE configuraton. Any idea? Br, Tamas On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: > Im using 2.6.32 on all my boxes ... > > One thing that has me thinking, are there any tweaks to get MSI-X working > best it can ? ( with proxmox ) > there seems to be a strong bias towards one CPU for all interrupts. > > I could be wrong, but its something I think ive seen, and didnt see any > clear suggestions on. > > Jay > > > > > > On 23 May 2013 01:12, Anthony Minessale wrote: > >> 2.6.25 or newer to get timerfd support. >> >> >> >> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >> >>> On 5/22/13, Anthony Minessale wrote: >>> > You should consider centos6 or debian stable. Make sure the host >>> kernel is >>> > very new to get maximum results. >>> >>> Tony, do you mean "very new kernel" means 3.2.xx kernel? >>> >>> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >>> possible. And that is what CentOS6 offers, too. >>> >>> However, I installed FS as openvz guest, it works fine for outgoing, >>> but not DNAT works for incoming connections even after throroughly >>> following >>> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >>> >>> Just my two cents. >>> >>> >>> >>> > >>> > >>> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >>> wrote: >>> > >>> >> Hello, >>> >> >>> >> Do you have any recommendations regarding how to set up correctly (for >>> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >>> >> tuneup >>> >> the system to be rock solid? >>> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container >>> with FS >>> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >>> >> something is not well configured around timers, however >>> mod_posix_timer >>> >> did >>> >> not help anything (running FS with -rp). We use our own bare metal and >>> >> can >>> >> reproduce those delatas eirher when only one VE is on the HW. >>> >> Maybe time to check out centos6 with openvz? >>> >> >>> >> Any idea, recommendation, experience can be very helpful. >>> >> >>> >> Regards, >>> >> Jalsot >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.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 >>> >> >>> >> >>> > >>> > >>> > -- >>> > Anthony Minessale II >>> > >>> > FreeSWITCH http://www.freeswitch.org/ >>> > ClueCon http://www.cluecon.com/ >>> > Twitter: http://twitter.com/FreeSWITCH_wire >>> > >>> > AIM: anthm >>> > MSN:anthony_minessale at hotmail.com >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> > IRC: irc.freenode.net #freeswitch >>> > >>> > FreeSWITCH Developer Conference >>> > sip:888 at conference.freeswitch.org >>> > googletalk:conf+888 at conference.freeswitch.org >>> > pstn:+19193869900 >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/890dacf8/attachment-0001.html From ml-ktk at netlabs.org Fri May 24 11:43:03 2013 From: ml-ktk at netlabs.org (Adrian Gschwend) Date: Fri, 24 May 2013 09:43:03 +0200 Subject: [Freeswitch-users] Google Talk XMPP Support In-Reply-To: References: Message-ID: <519F1A07.9060709@netlabs.org> On 21.05.13 15:51, Andrew Cassidy wrote: > I've heard rumors that XMPP support in Google Talk is being killed. I'm > guessing this means no more Google Talk in FreeSWITCH? some more details from the EFF https://www.eff.org/deeplinks/2013/05/google-abandons-open-standards-instant-messaging So server-to-server communication does not seem to work in Hangout. regards Adrian From ehermouet at bluetel.fr Fri May 24 11:54:58 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 24 May 2013 09:54:58 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Message-ID: <2e0378edf793985d2f24da75813d152b@bluetel.fr> I don't see rtp envent on wireshark. :'( Le 2013-05-23 18:06, Michael Jerris a ?crit?: > Have you looked at it to see if it is sending the dtmf? > > On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > >> you can found here my tcpdump file >> >> http://bluetelconnect.fr/tcpdump.log >> >> tks advance Michael >> >> >> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>> review of what you have posted so far. >>> >>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>> >>>> pcap. >>>> >>>> i send you the xml file and log in my previous email... because i >>>> see >>>> problem sometime... i'm sure i have error on my xml file. can you >>>> check >>>> it. ? >>>> >>>> tks >>>> >>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>> How do you use what? >>>>> >>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> how do you use it without interface ? it's server with only ssh >>>>>> access. >>>>>> tks >>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>> this log does not seem to have a complete call let alone any >>>>>>> attempt >>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>> said, >>>>>>> its >>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>> flowing >>>>>>> ? >>>>>>> >>>>>>> Mike >>>>>>> >>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> Yes >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>> >>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>> negotiation >>>>>>>>> settings? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ehermouet at bluetel.fr Fri May 24 12:43:37 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 24 May 2013 10:43:37 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Message-ID: i try incomming call and it's work... but i don't see dtmf on wireshark on incoming call i think it's the same on outbound... so my my prob is here again :'( Le 2013-05-23 18:06, Michael Jerris a ?crit?: > Have you looked at it to see if it is sending the dtmf? > > On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > >> you can found here my tcpdump file >> >> http://bluetelconnect.fr/tcpdump.log >> >> tks advance Michael >> >> >> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>> review of what you have posted so far. >>> >>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>> >>>> pcap. >>>> >>>> i send you the xml file and log in my previous email... because i >>>> see >>>> problem sometime... i'm sure i have error on my xml file. can you >>>> check >>>> it. ? >>>> >>>> tks >>>> >>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>> How do you use what? >>>>> >>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> how do you use it without interface ? it's server with only ssh >>>>>> access. >>>>>> tks >>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>> this log does not seem to have a complete call let alone any >>>>>>> attempt >>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>> said, >>>>>>> its >>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>> flowing >>>>>>> ? >>>>>>> >>>>>>> Mike >>>>>>> >>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> Yes >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>> >>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>> negotiation >>>>>>>>> settings? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From mehroz.ashraf85 at gmail.com Fri May 24 13:08:22 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 24 May 2013 02:08:22 -0700 (PDT) Subject: [Freeswitch-users] Centralized SIP directory Message-ID: <1369386502709-7591047.post@n2.nabble.com> Hi all, I have a DB server, storing SIP users and auth information. 2 freeswitch servers configured with mod_xml_curl , and an attempt of registration on both servers looks for SIP authentication from DB server. At the moment, i am defining 2 domains on DB server (FS-1 IP and FS-2 IP), and creating unique user , 2 times to address both domains. I want to get rid of creating 2 users for both domains. What is the best approach. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ehermouet at bluetel.fr Fri May 24 13:31:27 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 24 May 2013 11:31:27 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Message-ID: <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> After some hours i foudn rtp event with wireshark. RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) but no result on ivr outbound... Le 2013-05-23 18:06, Michael Jerris a ?crit?: > Have you looked at it to see if it is sending the dtmf? > > On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > >> you can found here my tcpdump file >> >> http://bluetelconnect.fr/tcpdump.log >> >> tks advance Michael >> >> >> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>> review of what you have posted so far. >>> >>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>> >>>> pcap. >>>> >>>> i send you the xml file and log in my previous email... because i >>>> see >>>> problem sometime... i'm sure i have error on my xml file. can you >>>> check >>>> it. ? >>>> >>>> tks >>>> >>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>> How do you use what? >>>>> >>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> how do you use it without interface ? it's server with only ssh >>>>>> access. >>>>>> tks >>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>> this log does not seem to have a complete call let alone any >>>>>>> attempt >>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>> said, >>>>>>> its >>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>> flowing >>>>>>> ? >>>>>>> >>>>>>> Mike >>>>>>> >>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> Yes >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>> >>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>> negotiation >>>>>>>>> settings? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From sertys at gmail.com Fri May 24 13:50:36 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 24 May 2013 11:50:36 +0200 Subject: [Freeswitch-users] Best performing codec on mobile/slow networks Message-ID: I've officially given up on AMR, due to it's licensing issues and sketchy supports for my mobile clients. I am looking for new options now as CsipSimple has a new codec pack supporting G.726.1, opus and codec2. Which do you think would perform best on mobile networks? And which has FS support for transcoding as well? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/ec13eac1/attachment.html From sertys at gmail.com Fri May 24 14:16:10 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 24 May 2013 12:16:10 +0200 Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <1369386502709-7591047.post@n2.nabble.com> References: <1369386502709-7591047.post@n2.nabble.com> Message-ID: Well, you're only serving the directory xml via the xml_curl. You need to make use of the Freeswitch's abiltity to use a custom database backend for it's registration and presence tables to make it foolproof. Check the core pgsql support or the odbc support in freeswitch for the listed things. What i have done however is to log the registration requests on the server side of the xml_curl servings and then create the bridge commands accordingly to where the user has last registered. It's a more crippled approach, but removes a point of failure for me (maintaining a pgsql server). On Fri, May 24, 2013 at 11:08 AM, mehroz wrote: > Hi all, > > I have a DB server, storing SIP users and auth information. > > 2 freeswitch servers configured with mod_xml_curl , and an attempt of > registration on both servers looks for SIP authentication from DB server. > At > the moment, i am defining 2 domains on DB server (FS-1 IP and FS-2 IP), and > creating unique user , 2 times to address both domains. > > I want to get rid of creating 2 users for both domains. What is the best > approach. > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047.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://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/20130524/2b23621c/attachment-0001.html From mehroz.ashraf85 at gmail.com Fri May 24 15:39:24 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 24 May 2013 04:39:24 -0700 (PDT) Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: References: <1369386502709-7591047.post@n2.nabble.com> Message-ID: <1369395564138-7591052.post@n2.nabble.com> Exactly, that is what i am doing. using ODBC with Mysql. My registration are saving in "sip_registration" table and i can get the IP location of each user (sip_host address). A perl script serving the purpose of fetching records from DB and knowing the IP location of callee and caller. But , i am trapped in "domain" structure of FS. I do not want to create two copies of a single user, like SIP number : 1000 with domain of FS IP-1 SIP number : 1000 with domain of FS IP-2 As both FS instance need to knwo if that user exists in their directory or not, which is depicted in terms of "domain" -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047p7591052.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Fri May 24 15:44:41 2013 From: steveu at coppice.org (Steve Underwood) Date: Fri, 24 May 2013 19:44:41 +0800 Subject: [Freeswitch-users] Best performing codec on mobile/slow networks In-Reply-To: References: Message-ID: <519F52A9.1000104@coppice.org> On 05/24/2013 05:50 PM, Daniel Ivanov wrote: > I've officially given up on AMR, due to it's licensing issues and > sketchy supports for my mobile clients. I am looking for new options > now as CsipSimple has a new codec pack supporting G.726.1, opus and > codec2. Which do you think would perform best on mobile networks? And > which has FS support for transcoding as well? > I think you means G.722.1 and G.726, rather than G.726. G.726 is narrowband and fairly high bit rate. Its not the greatest choice when you have packet loss. G.722.1 is wideband and fairly high bit rate. Its not the greatest choice when you have packet loss. codec2 is very low bit rate, which is good for some mobile users. The quality is good for the bit rate, but you can't expect high quality at such low bit rates. opus has lots of choices to trade off quality and bit rate. Its rather new, so not that widely supported, but it should become big over the next year or two. Steve From michel.brabants at gmail.com Fri May 24 15:53:41 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 24 May 2013 13:53:41 +0200 Subject: [Freeswitch-users] problem bridging lots of dtmf in a session Message-ID: Hello, we have a problem with dtmf not being bridged by freeswitch when there is lots of dtmf in the call. Freeswitch resides between an asterisk gateway (which sends lot of duplicated dtmf) and an ivr. When there is a lot of dtmf being bridged in a single call, we can see in a wireshark-trace that it stops bridging the dtmf at a certain moment. I can still see the dtmf being received in the freeswitch-logs, bit I'm not seeing it in the wireshark-trace towards the ivr. We're using freeswitch 1.2.7. Thanks, Michel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/e492a633/attachment.html From michel.brabants at gmail.com Fri May 24 15:55:08 2013 From: michel.brabants at gmail.com (Michel Brabants) Date: Fri, 24 May 2013 13:55:08 +0200 Subject: [Freeswitch-users] problem bridging lots of dtmf in a session In-Reply-To: References: Message-ID: info: auto_flush_during_bridge=false On Fri, May 24, 2013 at 1:53 PM, Michel Brabants wrote: > Hello, > > we have a problem with dtmf not being bridged by freeswitch when there is > lots of dtmf in the call. Freeswitch resides between an asterisk gateway > (which sends lot of duplicated dtmf) and an ivr. When there is a lot of > dtmf being bridged in a single call, we can see in a wireshark-trace that > it stops bridging the dtmf at a certain moment. I can still see the dtmf > being received in the freeswitch-logs, bit I'm not seeing it in the > wireshark-trace towards the ivr. > > We're using freeswitch 1.2.7. > > Thanks, > > Michel > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/e196a552/attachment.html From freeswitch-list at puzzled.xs4all.nl Fri May 24 15:59:14 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Fri, 24 May 2013 13:59:14 +0200 Subject: [Freeswitch-users] Best performing codec on mobile/slow networks In-Reply-To: References: Message-ID: <519F5612.3030207@puzzled.xs4all.nl> On 05/24/2013 11:50 AM, Daniel Ivanov wrote: > I've officially given up on AMR, due to it's licensing issues and > sketchy supports for my mobile clients. I am looking for new options now > as CsipSimple has a new codec pack supporting G.726.1, opus and codec2. > Which do you think would perform best on mobile networks? And which has > FS support for transcoding as well? Iirc last time I tested something on Android (don't recall the client) while on a 3G connection I got reasonable results with g.729, gsm and iLBC. On wifi with g.722. It depends on how good or crappy your mobile operator's network is and how busy the cell is that the phone is talking to. There are commercial g.729 licenses for FreeSWITCH. Info at: http://freeswitch.org/node/235 The other codecs you mentioned are all supported. For details see: http://wiki.freeswitch.org/wiki/Specsheet Regards, Patrick From ashish at nms.co.in Fri May 24 16:17:39 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 24 May 2013 17:47:39 +0530 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true Message-ID: Hi, I have a dialplan as follows: when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl does not get executed. Please help -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/aeca27fc/attachment.html From nneul at mst.edu Fri May 24 16:27:57 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 24 May 2013 07:27:57 -0500 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: Message-ID: <519F5CCD.8000609@mst.edu> I may be misunderstanding - but where are you causing it to ring a device? You've told it to internally answer the call, and then not do anything. There's no bridging to an actual extension. Only thing I see that would happen is it running perl/ash.pl, unclear if it would in term execute hook.pl when that script finished (I don't know what that behavior is expected to be). -- Nathan On 05/24/2013 07:17 AM, Ashish gautam wrote: > Hi, > > I have a dialplan as follows: > > > > > > > > > > > > > > when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl > does not get executed. > > Please help > -- > Ashish Gautam > > IVR Developer > > Nucleus Microsystems (Pvt.) Ltd. > > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From netcentrica at gmail.com Fri May 24 16:28:57 2013 From: netcentrica at gmail.com (Adam Raszynski) Date: Fri, 24 May 2013 14:28:57 +0200 Subject: [Freeswitch-users] Audio delay problem after upgrading to newest GIT version - increasing delay only on one leg In-Reply-To: References: Message-ID: Yes, it's related with Linksys PAP2T. The problem is that there are millions of this devices around the world :) I don't blame FreeSWITCH for that, changing codec params did the trick and now it works great as before. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/e4c5abc2/attachment.html From paul at cupis.co.uk Fri May 24 16:39:35 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 24 May 2013 13:39:35 +0100 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: Message-ID: <20130524123935.GA27353@eagle.cupis.co.uk> On Fri, May 24, 2013 at 05:47:39PM +0530, Ashish gautam wrote: > > when the called party does not pick up the phone or is busy, the > dialplan does not proceed and [3]hook.pl does not get executed. Perhaps "api_reporting_hook" would work for you in this use-case? Regards, From nneul at mst.edu Fri May 24 16:43:51 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 24 May 2013 07:43:51 -0500 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: <519F5CCD.8000609@mst.edu> Message-ID: <519F6087.3030706@mst.edu> I don't think that's going to do what you want... (May be wrong.) I think that continue_on_fail is only going to apply to the rules for the received call on this extension, not the received call on the outgoing leg. i.e. there are no dialplan rules in effect for the outgoing call that you initiated, and that's where the failure is occurring. For these dialplan rules, I think the only failure would be if your IVR (I assume that's was ash.pl is) didn't answer. Like I said, not certain of this, maybe some else can chime in, but I think you're going to have to handle that failure as a part of your originate on the outbound call. Something like putting originate {api_hangup_hook=perl hook.pl}sofia/..... Where you cause the call to take place. -- Nathan On 05/24/2013 07:37 AM, Ashish gautam wrote: > I am generating an outgoing call through mod_event_socket and then transferring it to this dialplan. > > On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger > wrote: > > I may be misunderstanding - but where are you causing it to ring a device? > > You've told it to internally answer the call, and then not do anything. There's no bridging to an actual extension. > > Only thing I see that would happen is it running perl/ash.pl , unclear if it would in term execute > hook.pl when that script finished (I don't know what that behavior is expected to be). > > -- Nathan > > > On 05/24/2013 07:17 AM, Ashish gautam wrote: > > Hi, > > I have a dialplan as follows: > > > > > > > > > > > > > > > when the called party does not pick up the phone or is busy, the dialplan does not proceed and hook.pl > > > does not get executed. > > Please help > -- > Ashish Gautam > > IVR Developer > > Nucleus Microsystems (Pvt.) Ltd. > > > > -- > ------------------------------__------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > > > > -- > Ashish Gautam > > IVR Developer > > Nucleus Microsystems (Pvt.) Ltd. > > Ph. 011 47574758 -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From ehermouet at bluetel.fr Fri May 24 17:16:34 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Fri, 24 May 2013 15:16:34 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> Message-ID: <6a6fa905293e71ff427cace19af47632@bluetel.fr> If i want to change to info, what i must remove and what i must add on my internal and externalx.xml except dtmf-type ? tks advance for your help Le 2013-05-23 18:06, Michael Jerris a ?crit?: > Have you looked at it to see if it is sending the dtmf? > > On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: > >> you can found here my tcpdump file >> >> http://bluetelconnect.fr/tcpdump.log >> >> tks advance Michael >> >> >> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>> review of what you have posted so far. >>> >>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>> >>>> pcap. >>>> >>>> i send you the xml file and log in my previous email... because i >>>> see >>>> problem sometime... i'm sure i have error on my xml file. can you >>>> check >>>> it. ? >>>> >>>> tks >>>> >>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>> How do you use what? >>>>> >>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> how do you use it without interface ? it's server with only ssh >>>>>> access. >>>>>> tks >>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>> this log does not seem to have a complete call let alone any >>>>>>> attempt >>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>> said, >>>>>>> its >>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>> flowing >>>>>>> ? >>>>>>> >>>>>>> Mike >>>>>>> >>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> Yes >>>>>>>> >>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>> >>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>> negotiation >>>>>>>>> settings? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From sertys at gmail.com Fri May 24 17:34:59 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Fri, 24 May 2013 15:34:59 +0200 Subject: [Freeswitch-users] Best performing codec on mobile/slow networks In-Reply-To: <519F5612.3030207@puzzled.xs4all.nl> References: <519F5612.3030207@puzzled.xs4all.nl> Message-ID: I was using g729 until recently, but regis gave up on licensing issues and removed the g729 from the stock builds. There is a paid codec from securax though, but im not advising customers to buy it. On May 24, 2013 3:02 PM, "Patrick Lists" wrote: > On 05/24/2013 11:50 AM, Daniel Ivanov wrote: > > I've officially given up on AMR, due to it's licensing issues and > > sketchy supports for my mobile clients. I am looking for new options now > > as CsipSimple has a new codec pack supporting G.726.1, opus and codec2. > > Which do you think would perform best on mobile networks? And which has > > FS support for transcoding as well? > > Iirc last time I tested something on Android (don't recall the client) > while on a 3G connection I got reasonable results with g.729, gsm and > iLBC. On wifi with g.722. It depends on how good or crappy your mobile > operator's network is and how busy the cell is that the phone is talking > to. > > There are commercial g.729 licenses for FreeSWITCH. Info at: > http://freeswitch.org/node/235 > > The other codecs you mentioned are all supported. For details see: > http://wiki.freeswitch.org/wiki/Specsheet > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/1dcd8f4b/attachment.html From j_mj at aol.com Fri May 24 18:27:47 2013 From: j_mj at aol.com (John M) Date: Fri, 24 May 2013 10:27:47 -0400 (EDT) Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <1369395564138-7591052.post@n2.nabble.com> References: <1369386502709-7591047.post@n2.nabble.com> <1369395564138-7591052.post@n2.nabble.com> Message-ID: <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> You might be looking at it the wrong way... You don't need to entries of the user, you need a single entry but with multiple domain values. So the curl lookup selects from the db where user = xx and domain LIKE %xx% something like that -Jm -----Original Message----- From: mehroz To: freeswitch-users Sent: Fri, May 24, 2013 9:46 pm Subject: Re: [Freeswitch-users] Centralized SIP directory Exactly, that is what i am doing. using ODBC with Mysql. My registration are saving in "sip_registration" table and i can get the IP location of each user (sip_host address). A perl script serving the purpose of fetching records from DB and knowing the IP location of callee and caller. But , i am trapped in "domain" structure of FS. I do not want to create two copies of a single user, like SIP number : 1000 with domain of FS IP-1 SIP number : 1000 with domain of FS IP-2 As both FS instance need to knwo if that user exists in their directory or not, which is depicted in terms of "domain" -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047p7591052.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://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/20130524/de8e4e1f/attachment.html From flavio at voffice.com.br Fri May 24 00:38:17 2013 From: flavio at voffice.com.br (Flavio Goncalves) Date: Thu, 23 May 2013 17:38:17 -0300 Subject: [Freeswitch-users] Strange issue with late negotiation Message-ID: Hello, I'm having a really strange issue on billing related to late negotiation. The call end up hanged with no BYE. There is a dialer sending thousands of calls through and OpenSIPS to a termination based on FreeSwitch. I have a SIP call flow as below. Some times during the day, when the volume is high, the UAC drops some 200Ok and FS send a reinvite in the middle of the Initial transaction. ----INVITE --------------- Proxy --INVITE ---------------> FS <----200 OK -------------- Proxy <---200 OK- ------------- FS <--REINVITE--------------- Proxy <-REINVITE--------------- FS ---481 leg does not exit-> Proxy ---481------------------> FS <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS ----CANCEL ---------------> Proxy ---CANCEL --------------> FS <---200 Ok ---------------- Proxy <----200 Ok ------------- FS FS is sending a reinvite before ACK comes from the client. I have two questions: 1) Is it valid to send a reinvite in the middle of an existing transaction? According to the RFC3261 Section 14. Note that a UAC MUST NOT initiate a new INVITE transaction within a dialog while another INVITE transaction is in progress in either direction. 1. If there is an ongoing INVITE client transaction, the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. 2. If there is an ongoing INVITE server transaction, the TU MUST wait until the transaction reaches the confirmed or terminated state before initiating the new INVITE. But it is confusing because just below it says the opposite. However, a UA MAY initiate a regular transaction while an INVITE transaction is in progress. A UA MAY also initiate an INVITE transaction while a regular transaction is in progress. 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK? If a UAS generates a 2xx response and never receives an ACK, it SHOULD generate a BYE to terminate the dialog. Best regards, Flavio E. Goncalves -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/f1d347f2/attachment-0001.html From cgoudie at getjive.com Fri May 24 03:22:03 2013 From: cgoudie at getjive.com (Clinton Goudie-Nice) Date: Thu, 23 May 2013 17:22:03 -0600 Subject: [Freeswitch-users] ESL using bridge app doesn't return which gateway was used Message-ID: When you make a bridge command using esl, where you specify multiple gateways or sip dials separated by or bars, you can't figure out which gateway was used. For example, if you bridge to something like this: sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000 The call could be bridged to either GW2 or GW1. When the CHANNEL_BRIDGE event is returned, you can see the original string in variable_current_application_data, and you may be able to infer the destination based on IP address, but nothing clearly says what gateway is used. If you turn on the all events firehose, you can see the CHANNEL_CREATE event come over the socket, and it does contain variable_sip_gateway_name with the actual name of the gateway, however I can't devise a way to access that data using the org.freeswitch.esl.client library, and even if I could, I still don't want all events for this system. Is it possible to get this information returned in any meaningful way through the ESL layer, either by an api command to query, or the setting of a variable that will give me back which gateway a bridge was performed through? If none of that is possible, this sounds worthy of filing a bug to return the variable_sip_gateway_name in the CHANNEL_BRIDGE event. Thanks for the help, Clint -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130523/6787153c/attachment.html From jkhan6722 at gmail.com Fri May 24 10:59:31 2013 From: jkhan6722 at gmail.com (juned) Date: Thu, 23 May 2013 23:59:31 -0700 (PDT) Subject: [Freeswitch-users] voicemail is not working Message-ID: <1369378771206-7591043.post@n2.nabble.com> Hi All, I am newbie to FS. so as a startup i have installed FS in mu local system to test out the basic functionality and features. so i have registered default users 1000 and 1001 in softphone( twinkle ). registration was successful and calls was also fine but when i tried to check voicemail then it didn't worked. what i did to test it out voicemail is, did call to 1001 and let it rang so after 30 second if no answer is there then voicemail will be activated but calls are released after 30 seconds. As per documentation i came to know that by default voicemail is activated in default extensions is it so ? or i am missing something. Please point me to right direction, i want to have such a dialplan in which user can leave voicemail in cases of busy,unavailable and not answering. Thanks & Regards Juned -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/voicemail-is-not-working-tp7591043.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shahzad.bhatti at g-r-v.com Fri May 24 18:51:31 2013 From: shahzad.bhatti at g-r-v.com (Shahzad Bhatti) Date: Fri, 24 May 2013 19:51:31 +0500 Subject: [Freeswitch-users] mod_cdr_csv is not working! Message-ID: i open a jira and here is a link of that http://jira.freeswitch.org/browse/FS-5468 Shahzad Bhatti On Wed, May 22, 2013 at 8:50 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: mod_cdr_csv is not working! (Michael Collins) > 2. Re: One Way Audio (Michael Collins) > 3. Re: spam>spam>Re: One Way Audio (Moishe Grunstein) > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Wed, 22 May 2013 08:39:26 -0700 > Subject: Re: [Freeswitch-users] mod_cdr_csv is not working! > Can you reproduce this on latest HEAD? If so, gather details and open a > Jira. > > -MC > > > On Wed, May 22, 2013 at 8:00 AM, Shahzad Bhatti wrote: > >> Hi, >> >> after loading *mod_cdr_csv, *when i originate test calls the cdr is not >> appended in Master.csv file most of the time but sometime cdr is appended >> in the Master.csv that show that mod_cdr_csv is configured. Also when i try >> to use *cdr_csv rotate* command on fs_cli it also not work. even i also >> try to do using the perl script example available on following url >> >> >> https://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_Perl_Script_for_CDR_into_MySQL >> >> i want to know why this is happening on my server and how i can fix these >> issues >> >> 1. cdrs are not updated in the csv file; >> 2. rotate command is not working; >> >> >> any reply is highly appreciated; >> >> Regards >> >> Shahzad Bhatti >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/248819b1/attachment.html From mehroz.ashraf85 at gmail.com Fri May 24 18:54:58 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Fri, 24 May 2013 07:54:58 -0700 (PDT) Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> References: <1369386502709-7591047.post@n2.nabble.com> <1369395564138-7591052.post@n2.nabble.com> <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> Message-ID: <1369407298402-7591066.post@n2.nabble.com> Alright, and how do we tell it to support multiple domain with a single user? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047p7591066.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Fri May 24 18:58:53 2013 From: mike at jerris.com (Michael Jerris) Date: Fri, 24 May 2013 10:58:53 -0400 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <2a967567116b62bd991f9eb2ae525cb5@bluetel.fr> <012701ce525a$f59c2b70$e0d48250$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> Message-ID: <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> if you see the rtp events going out but you don't see it having any affect, try asking your provider? Mike On May 24, 2013, at 5:31 AM, ehermouet at bluetel.fr wrote: > After some hours i foudn rtp event with wireshark. > > RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) > but no result on ivr outbound... > > > > > Le 2013-05-23 18:06, Michael Jerris a ?crit : >> Have you looked at it to see if it is sending the dtmf? >> >> On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: >> >>> you can found here my tcpdump file >>> >>> http://bluetelconnect.fr/tcpdump.log >>> >>> tks advance Michael >>> >>> >>> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>>> review of what you have posted so far. >>>> >>>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>>> >>>>> pcap. >>>>> >>>>> i send you the xml file and log in my previous email... because i >>>>> see >>>>> problem sometime... i'm sure i have error on my xml file. can you >>>>> check >>>>> it. ? >>>>> >>>>> tks >>>>> >>>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>>> How do you use what? >>>>>> >>>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>>> >>>>>>> how do you use it without interface ? it's server with only ssh >>>>>>> access. >>>>>>> tks >>>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>>> this log does not seem to have a complete call let alone any >>>>>>>> attempt >>>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>>> said, >>>>>>>> its >>>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>>> flowing >>>>>>>> ? >>>>>>>> >>>>>>>> Mike >>>>>>>> >>>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>>> >>>>>>>>> Yes >>>>>>>>> >>>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>>> >>>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>>> negotiation >>>>>>>>>> settings? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From ehermouet at bluetel.fr Fri May 24 19:19:57 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Fri, 24 May 2013 17:19:57 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> Message-ID: <6fc85078-e753-4807-b37d-bf625fea8f4c@email.android.com> I ask him. He said info will be better than 2833. But i try in info and the same result Michael Jerris a ?crit?: >if you see the rtp events going out but you don't see it having any >affect, try asking your provider? > >Mike > >On May 24, 2013, at 5:31 AM, ehermouet at bluetel.fr wrote: > >> After some hours i foudn rtp event with wireshark. >> >> RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) >> but no result on ivr outbound... >> >> >> >> >> Le 2013-05-23 18:06, Michael Jerris a ?crit : >>> Have you looked at it to see if it is sending the dtmf? >>> >>> On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: >>> >>>> you can found here my tcpdump file >>>> >>>> http://bluetelconnect.fr/tcpdump.log >>>> >>>> tks advance Michael >>>> >>>> >>>> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>>>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>>>> review of what you have posted so far. >>>>> >>>>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> pcap. >>>>>> >>>>>> i send you the xml file and log in my previous email... because i >>>>>> see >>>>>> problem sometime... i'm sure i have error on my xml file. can you >>>>>> check >>>>>> it. ? >>>>>> >>>>>> tks >>>>>> >>>>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>>>> How do you use what? >>>>>>> >>>>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> how do you use it without interface ? it's server with only ssh >>>>>>>> access. >>>>>>>> tks >>>>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>>>> this log does not seem to have a complete call let alone any >>>>>>>>> attempt >>>>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>>>> said, >>>>>>>>> its >>>>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>>>> flowing >>>>>>>>> ? >>>>>>>>> >>>>>>>>> Mike >>>>>>>>> >>>>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>>>> >>>>>>>>>> Yes >>>>>>>>>> >>>>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>>>> >>>>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>>>> negotiation >>>>>>>>>>> settings? >>> >>> >>> >_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/37c26156/attachment-0001.html From j_mj at aol.com Fri May 24 19:34:53 2013 From: j_mj at aol.com (John M) Date: Fri, 24 May 2013 11:34:53 -0400 (EDT) Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <1369407298402-7591066.post@n2.nabble.com> References: <1369386502709-7591047.post@n2.nabble.com> <1369395564138-7591052.post@n2.nabble.com> <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> <1369407298402-7591066.post@n2.nabble.com> Message-ID: <8D026AB5F145D0E-18C4-15C15@webmail-m236.sysops.aol.com> I'm assuming by 'it' you mean freeswitch. I expect you don't need to, you send a curl request off to the central server from FS01 checking the credentials of the user trying to log in. On the central server it currently does a db lookup for the user that is stored for the domain on the FS01 server. I assume that will be in a column 'domain' or some such. Just expand that column domain to store the domains of both servers and when the lookup comes in match against the value with the mysql LIKE function instead of = Or it's possible that I completely don't understand what you are trying to do. -Jm -----Original Message----- From: mehroz To: freeswitch-users Sent: Sat, May 25, 2013 1:04 am Subject: Re: [Freeswitch-users] Centralized SIP directory Alright, and how do we tell it to support multiple domain with a single user? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047p7591066.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://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/20130524/73850920/attachment.html From mbodbg at gmx.net Fri May 24 19:44:24 2013 From: mbodbg at gmx.net (mbo) Date: Fri, 24 May 2013 17:44:24 +0200 Subject: [Freeswitch-users] Freeswitch versioning / tagging /releasemanagement Message-ID: On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I can see that there are many tags with a higher version, but older tag date. v1.3.0 1.3.0 release v1.3.1 Tagging version 1.3.1 v1.3.10 tag v1.3.10 v1.3.11 tag v1.3.11 v1.3.12 Retag v1.3.12 v1.3.13 tag v1.3.13 v1.3.14 tag v1.3.14 v1.3.15 tag v1.3.15 v1.3.16 tag v1.3.16 v1.3.17-final tag v1.3.17-final v1.3.2 Tagging version 1.3.2 v1.3.3 Tagging 1.3.3 v1.3.4 release v1.3.4 v1.3.5 release v1.3.5 v1.3.6 tag v1.3.6 v1.3.7 tag v1.3.7 v1.3.8 tag v1.3.8 v1.3.9 tag v1.3.9 v1.5.0 tag v1.5.0 What about those tags? How is the release management organized? Thanks Markus From paul at cupis.co.uk Fri May 24 19:54:10 2013 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 24 May 2013 16:54:10 +0100 Subject: [Freeswitch-users] Freeswitch versioning / tagging /releasemanagement In-Reply-To: References: Message-ID: <20130524155410.GA31606@eagle.cupis.co.uk> On Fri, May 24, 2013 at 05:44:24PM +0200, mbo wrote: > On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I look into git (git tag -l -n1), I can see that there are many tags with a higher version, but older tag date. > What about those tags? How is the release management organized? 1. is a stable release. 1. is a development release. I'd suggest using the v1.2.stable branch as this is the most "release" version currently. Regards, From krice at freeswitch.org Fri May 24 19:55:46 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 24 May 2013 10:55:46 -0500 Subject: [Freeswitch-users] Freeswitch versioning / tagging /releasemanagement In-Reply-To: Message-ID: 1.X.Y... Where X Is Even, that's Stable Where X is Odd that's Dev... X Odd dev branch feeds X-1 Stable Branch... On 5/24/13 10:44 AM, "mbo" wrote: > On the wiki I can see that the latest stable freeswitch release is 1.2.9. If I > look into git (git tag -l -n1), I can see that there are many tags with a > higher version, but older tag date. > > v1.3.0 1.3.0 release > v1.3.1 Tagging version 1.3.1 > v1.3.10 tag v1.3.10 > v1.3.11 tag v1.3.11 > v1.3.12 Retag v1.3.12 > v1.3.13 tag v1.3.13 > v1.3.14 tag v1.3.14 > v1.3.15 tag v1.3.15 > v1.3.16 tag v1.3.16 > v1.3.17-final tag v1.3.17-final > v1.3.2 Tagging version 1.3.2 > v1.3.3 Tagging 1.3.3 > v1.3.4 release v1.3.4 > v1.3.5 release v1.3.5 > v1.3.6 tag v1.3.6 > v1.3.7 tag v1.3.7 > v1.3.8 tag v1.3.8 > v1.3.9 tag v1.3.9 > v1.5.0 tag v1.5.0 > > What about those tags? How is the release management organized? > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From bdfoster at davri.com Fri May 24 20:01:04 2013 From: bdfoster at davri.com (Brian Foster) Date: Fri, 24 May 2013 12:01:04 -0400 Subject: [Freeswitch-users] Freeswitch versioning / tagging /releasemanagement In-Reply-To: <20130524155410.GA31606@eagle.cupis.co.uk> References: <20130524155410.GA31606@eagle.cupis.co.uk> Message-ID: I'd like to add that everything v1.3.0 is not stable. - BDF On May 24, 2013 11:59 AM, "Paul Cupis" wrote: > On Fri, May 24, 2013 at 05:44:24PM +0200, mbo wrote: > > On the wiki I can see that the latest stable freeswitch release is > 1.2.9. If I look into git (git tag -l -n1), I can see that there are many > tags with a higher version, but older tag date. > > > What about those tags? How is the release management organized? > > 1. is a stable release. > 1. is a development release. > > I'd suggest using the v1.2.stable branch as this is the most "release" > version currently. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/ed222313/attachment.html From anthony.minessale at gmail.com Fri May 24 21:00:39 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 May 2013 12:00:39 -0500 Subject: [Freeswitch-users] Strange issue with late negotiation In-Reply-To: References: Message-ID: Its probably not a re-invite but a retransmission of the same one because it never got the response. That RFC is trying to say you can't send another invite while you are waiting for a reply to the current one but you can send other things like info. The problem is when one side sends invite and the other end replies but the sender does not get that reply, they will keep sending the invite until it gives up. On Thu, May 23, 2013 at 3:38 PM, Flavio Goncalves wrote: > Hello, > > I'm having a really strange issue on billing related to late negotiation. > The call end up hanged with no BYE. > > There is a dialer sending thousands of calls through and OpenSIPS to a > termination based on FreeSwitch. I have a SIP call flow as below. Some > times during the day, when the volume is high, the UAC drops some 200Ok > and FS send a reinvite in the middle of the Initial transaction. > > ----INVITE --------------- Proxy --INVITE ---------------> FS > <----200 OK -------------- Proxy <---200 OK- ------------- FS > <--REINVITE--------------- Proxy <-REINVITE--------------- FS > ---481 leg does not exit-> Proxy ---481------------------> FS > <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS > ----CANCEL ---------------> Proxy ---CANCEL --------------> FS > <---200 Ok ---------------- Proxy <----200 Ok ------------- FS > > FS is sending a reinvite before ACK comes from the client. > > I have two questions: > > 1) Is it valid to send a reinvite in the middle of an existing transaction? > > According to the RFC3261 Section 14. > > Note that a UAC MUST NOT initiate a new INVITE transaction within a > dialog while another INVITE transaction is in progress in either > direction. > > 1. If there is an ongoing INVITE client transaction, the TU MUST > wait until the transaction reaches the completed or terminated > state before initiating the new INVITE. > > 2. If there is an ongoing INVITE server transaction, the TU MUST > wait until the transaction reaches the confirmed or terminated > state before initiating the new INVITE. > > > But it is confusing because just below it says the opposite. > > However, a UA MAY initiate a regular transaction while an INVITE > transaction is in progress. A UA MAY also initiate an INVITE > transaction while a regular transaction is in progress. > > > 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK? > > If a UAS generates a 2xx response and never receives an ACK, it > SHOULD generate a BYE to terminate the dialog. > > > Best regards, > > Flavio E. Goncalves > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/ccf0f705/attachment-0001.html From grcamauer at gmail.com Fri May 24 21:25:06 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 24 May 2013 14:25:06 -0300 Subject: [Freeswitch-users] Assertion when Transcoding Message-ID: I am getting the following assertion when attempting to record a G.729 call: freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. The system has a Sangoma D100 card installed. The call is established using the G.729 codec, and after playing some prompts, I am attempting to record an answer: freeswitch at 192.168.9.151:5000 at internal> show codecs type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,GSM,mod_spandsp codec,LPC-10,mod_spandsp codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,Sangoma G729,mod_sangoma_codec 18 total. . . . 2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File [/mnt/TEMP/beep-7.G729] 8000hz 2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] 2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] 2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] 2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY] freeswitch at 192.168.9.151:5000 at internal> freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion `channel != ((void *)0)' failed. FreeSwitch dies after this. Note that this does not happen if I restrict everything to G711 (ULAW). Any thoughts? -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/e7bd2771/attachment.html From anthony.minessale at gmail.com Fri May 24 21:38:19 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 May 2013 12:38:19 -0500 Subject: [Freeswitch-users] Assertion when Transcoding In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs On Fri, May 24, 2013 at 12:25 PM, Guillermo Ruiz Camauer < grcamauer at gmail.com> wrote: > I am getting the following assertion when attempting to record a G.729 > call: > > freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: > Assertion `channel != ((void *)0)' failed. > > The system has a Sangoma D100 card installed. The call is established > using the G.729 codec, and after playing some prompts, I am attempting to > record an answer: > > freeswitch at 192.168.9.151:5000 at internal> show codecs > type,name,ikey > codec,ADPCM (IMA),mod_spandsp > codec,G.711 alaw,CORE_PCM_MODULE > codec,G.711 ulaw,CORE_PCM_MODULE > codec,G.722,mod_spandsp > codec,G.726 16k,mod_spandsp > codec,G.726 16k (AAL2),mod_spandsp > codec,G.726 24k,mod_spandsp > codec,G.726 24k (AAL2),mod_spandsp > codec,G.726 32k,mod_spandsp > codec,G.726 32k (AAL2),mod_spandsp > codec,G.726 40k,mod_spandsp > codec,G.726 40k (AAL2),mod_spandsp > codec,GSM,mod_spandsp > codec,LPC-10,mod_spandsp > codec,PROXY PASS-THROUGH,CORE_PCM_MODULE > codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE > codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE > codec,Sangoma G729,mod_sangoma_codec > 18 total. > . > . > . > 2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 > sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] > 2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 > sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] > 2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 > sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] > 2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File > [/mnt/TEMP/beep-7.G729] 8000hz > 2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 > sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] > 2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 > sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] > 2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 > sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] > 2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 > sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY] > freeswitch at 192.168.9.151:5000 at internal> freeswitch: > src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion > `channel != ((void *)0)' failed. > > FreeSwitch dies after this. Note that this does not happen if I restrict > everything to G711 (ULAW). > > Any thoughts? > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/fc646fb1/attachment.html From grcamauer at gmail.com Fri May 24 22:31:30 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 24 May 2013 15:31:30 -0300 Subject: [Freeswitch-users] Assertion when Transcoding In-Reply-To: References: Message-ID: Filed JIRA FS-5472 with a backtrace. On Fri, May 24, 2013 at 2:38 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > On Fri, May 24, 2013 at 12:25 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I am getting the following assertion when attempting to record a G.729 >> call: >> >> freeswitch: src/switch_channel.c:832: switch_channel_get_variable_dup: >> Assertion `channel != ((void *)0)' failed. >> >> The system has a Sangoma D100 card installed. The call is established >> using the G.729 codec, and after playing some prompts, I am attempting to >> record an answer: >> >> freeswitch at 192.168.9.151:5000 at internal> show codecs >> type,name,ikey >> codec,ADPCM (IMA),mod_spandsp >> codec,G.711 alaw,CORE_PCM_MODULE >> codec,G.711 ulaw,CORE_PCM_MODULE >> codec,G.722,mod_spandsp >> codec,G.726 16k,mod_spandsp >> codec,G.726 16k (AAL2),mod_spandsp >> codec,G.726 24k,mod_spandsp >> codec,G.726 24k (AAL2),mod_spandsp >> codec,G.726 32k,mod_spandsp >> codec,G.726 32k (AAL2),mod_spandsp >> codec,G.726 40k,mod_spandsp >> codec,G.726 40k (AAL2),mod_spandsp >> codec,GSM,mod_spandsp >> codec,LPC-10,mod_spandsp >> codec,PROXY PASS-THROUGH,CORE_PCM_MODULE >> codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE >> codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE >> codec,Sangoma G729,mod_sangoma_codec >> 18 total. >> . >> . >> . >> 2013-05-24 12:33:31.071446 [ALERT] switch_core_session.c:2739 >> sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] >> 2013-05-24 12:33:31.071446 [ALERT] switch_ivr.c:650 >> sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] >> 2013-05-24 12:33:31.111446 [ALERT] switch_core_session.c:2724 >> sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] >> 2013-05-24 12:33:31.111446 [INFO] mod_native_file.c:94 Opening File >> [/mnt/TEMP/beep-7.G729] 8000hz >> 2013-05-24 12:33:31.231447 [ALERT] switch_core_session.c:2739 >> sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC_COMPLETE] >> 2013-05-24 12:33:31.231447 [ALERT] switch_ivr.c:650 >> sofia/crossfonetrunk/52764439 receive message [AUDIO_SYNC] >> 2013-05-24 12:33:31.271451 [ALERT] switch_core_session.c:2724 >> sofia/crossfonetrunk/52764439 receive message [APPLICATION_EXEC] >> 2013-05-24 12:33:31.291450 [ALERT] switch_core_io.c:446 >> sofia/crossfonetrunk/52764439 receive message [TRANSCODING_NECESSARY] >> freeswitch at 192.168.9.151:5000 at internal> freeswitch: >> src/switch_channel.c:832: switch_channel_get_variable_dup: Assertion >> `channel != ((void *)0)' failed. >> >> FreeSwitch dies after this. Note that this does not happen if I restrict >> everything to G711 (ULAW). >> >> Any thoughts? >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/5e0dc851/attachment-0001.html From mbodbg at gmx.net Fri May 24 22:51:02 2013 From: mbodbg at gmx.net (mbo) Date: Fri, 24 May 2013 20:51:02 +0200 Subject: [Freeswitch-users] WebRTC Message-ID: I've seen there have been a couple of discussions about WebRTC support earlier this year. Are there any news on that? Thanks Markus From flavio at sippulse.com Fri May 24 23:27:36 2013 From: flavio at sippulse.com (Flavio Goncalves) Date: Fri, 24 May 2013 16:27:36 -0300 Subject: [Freeswitch-users] Strange issue with late negotiation In-Reply-To: References: Message-ID: Hi Anthony, It is not a retransmission, the CSEQ is much higher and the branch is different. It is actually another transaction. The CSEQ of the first INVITE is 103, the second INVITE is something bigger than 400000. The biggest issue here for us is not receiving the BYE. We had built a mechanism in the proxy to send a BYE if not receiving the ACK in 5 seconds, but the ACK for the REINVITE disables it. We will change the mechanism to accept only the ACK for the first transaction, but anyway, I believe FS should have sent the BYE. If this can be fixed (if actually is broken), it would help to eliminate many billing issues when late negotiation is involved and the client stops answering (caused by network failures or client overload). Best regards, Flavio E. Goncalves CTO - SipPulse Routing and Billing Solutions for SIP Phone: +55 48-3025-8590, +1 248-688-0960 VoIP: SIP:flavio at opensips.org,Skype:flaviogoncalves1 Linkedin: www.linkedin.com/in/flavioegoncalves Twitter: www.twitter.com/asteriskguide 2013/5/24 Anthony Minessale > Its probably not a re-invite but a retransmission of the same one because > it never got the response. > > That RFC is trying to say you can't send another invite while you are > waiting for a reply to the current one but you can send other things like > info. The problem is when one side sends invite and the other end replies > but the sender does not get that reply, they will keep sending the invite > until it gives up. > > > On Thu, May 23, 2013 at 3:38 PM, Flavio Goncalves wrote: > >> Hello, >> >> I'm having a really strange issue on billing related to late negotiation. >> The call end up hanged with no BYE. >> >> There is a dialer sending thousands of calls through and OpenSIPS to a >> termination based on FreeSwitch. I have a SIP call flow as below. Some >> times during the day, when the volume is high, the UAC drops some 200Ok >> and FS send a reinvite in the middle of the Initial transaction. >> >> ----INVITE --------------- Proxy --INVITE ---------------> FS >> <----200 OK -------------- Proxy <---200 OK- ------------- FS >> <--REINVITE--------------- Proxy <-REINVITE--------------- FS >> ---481 leg does not exit-> Proxy ---481------------------> FS >> <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS >> ----CANCEL ---------------> Proxy ---CANCEL --------------> FS >> <---200 Ok ---------------- Proxy <----200 Ok ------------- FS >> >> FS is sending a reinvite before ACK comes from the client. >> >> I have two questions: >> >> 1) Is it valid to send a reinvite in the middle of an existing >> transaction? >> >> According to the RFC3261 Section 14. >> >> Note that a UAC MUST NOT initiate a new INVITE transaction within a >> dialog while another INVITE transaction is in progress in either >> direction. >> >> 1. If there is an ongoing INVITE client transaction, the TU MUST >> wait until the transaction reaches the completed or terminated >> state before initiating the new INVITE. >> >> 2. If there is an ongoing INVITE server transaction, the TU MUST >> wait until the transaction reaches the confirmed or terminated >> state before initiating the new INVITE. >> >> >> But it is confusing because just below it says the opposite. >> >> However, a UA MAY initiate a regular transaction while an INVITE >> transaction is in progress. A UA MAY also initiate an INVITE >> transaction while a regular transaction is in progress. >> >> >> 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the >> ACK? >> >> If a UAS generates a 2xx response and never receives an ACK, it >> SHOULD generate a BYE to terminate the dialog. >> >> >> Best regards, >> >> Flavio E. Goncalves >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/25c2f2c4/attachment.html From krice at freeswitch.org Fri May 24 23:28:57 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 24 May 2013 14:28:57 -0500 Subject: [Freeswitch-users] WebRTC In-Reply-To: Message-ID: Its coming sooner then later... Watch for info On 5/24/13 1:51 PM, "mbo" wrote: > I've seen there have been a couple of discussions about WebRTC support earlier > this year. Are there any news on that? > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From mike.burlingame at me.com Fri May 24 23:39:04 2013 From: mike.burlingame at me.com (Mike Burlingame) Date: Fri, 24 May 2013 12:39:04 -0700 Subject: [Freeswitch-users] Strange issue with late negotiation In-Reply-To: References: Message-ID: <8BDBDC72-CEC5-40AD-A6A6-01E1D505BB2D@me.com> Not sure this will resolve or help your issue - it addressed some of my issues with FS not waiting for the A-LEG to ACK before moving on https://wiki.freeswitch.org/wiki/Variable_sip_wait_for_aleg_ack On May 24, 2013, at 12:27 PM, Flavio Goncalves wrote: > Hi Anthony, > > It is not a retransmission, the CSEQ is much higher and the branch is different. It is actually another transaction. The CSEQ of the first INVITE is 103, the second INVITE is something bigger than 400000. The biggest issue here for us is not receiving the BYE. We had built a mechanism in the proxy to send a BYE if not receiving the ACK in 5 seconds, but the ACK for the REINVITE disables it. We will change the mechanism to accept only the ACK for the first transaction, but anyway, I believe FS should have sent the BYE. If this can be fixed (if actually is broken), it would help to eliminate many billing issues when late negotiation is involved and the client stops answering (caused by network failures or client overload). > > Best regards, > > > Flavio E. Goncalves > CTO - SipPulse Routing and Billing Solutions for SIP > Phone: +55 48-3025-8590, +1 248-688-0960 > VoIP: SIP:flavio at opensips.org,Skype:flaviogoncalves1 > Linkedin: www.linkedin.com/in/flavioegoncalves > Twitter: www.twitter.com/asteriskguide > > > > 2013/5/24 Anthony Minessale > Its probably not a re-invite but a retransmission of the same one because it never got the response. > > That RFC is trying to say you can't send another invite while you are waiting for a reply to the current one but you can send other things like info. The problem is when one side sends invite and the other end replies but the sender does not get that reply, they will keep sending the invite until it gives up. > > > On Thu, May 23, 2013 at 3:38 PM, Flavio Goncalves wrote: > Hello, > > I'm having a really strange issue on billing related to late negotiation. The call end up hanged with no BYE. > > There is a dialer sending thousands of calls through and OpenSIPS to a termination based on FreeSwitch. I have a SIP call flow as below. Some times during the day, when the volume is high, the UAC drops some 200Ok and FS send a reinvite in the middle of the Initial transaction. > > ----INVITE --------------- Proxy --INVITE ---------------> FS > <----200 OK -------------- Proxy <---200 OK- ------------- FS > <--REINVITE--------------- Proxy <-REINVITE--------------- FS > ---481 leg does not exit-> Proxy ---481------------------> FS > <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS > ----CANCEL ---------------> Proxy ---CANCEL --------------> FS > <---200 Ok ---------------- Proxy <----200 Ok ------------- FS > > FS is sending a reinvite before ACK comes from the client. > > I have two questions: > > 1) Is it valid to send a reinvite in the middle of an existing transaction? > > According to the RFC3261 Section 14. > > Note that a UAC MUST NOT initiate a new INVITE transaction within a > dialog while another INVITE transaction is in progress in either > direction. > > 1. If there is an ongoing INVITE client transaction, the TU MUST > wait until the transaction reaches the completed or terminated > state before initiating the new INVITE. > > 2. If there is an ongoing INVITE server transaction, the TU MUST > wait until the transaction reaches the confirmed or terminated > state before initiating the new INVITE. > > But it is confusing because just below it says the opposite. > > However, a UA MAY initiate a regular transaction while an INVITE > transaction is in progress. A UA MAY also initiate an INVITE > transaction while a regular transaction is in progress. > > 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK? > > If a UAS generates a 2xx response and never receives an ACK, it > SHOULD generate a BYE to terminate the dialog. > > Best regards, > > Flavio E. Goncalves > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/51c0daa3/attachment-0001.html From anthony.minessale at gmail.com Sat May 25 00:54:16 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 May 2013 15:54:16 -0500 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: Message-ID: Cluecon will have a bunch of of presentations related to it and we will make considerations then on how it could relate to the project. We are focused on our split of 1.2 / 1.4 right now. On May 24, 2013 1:55 PM, "mbo" wrote: > I've seen there have been a couple of discussions about WebRTC support > earlier this year. Are there any news on that? > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/f79cacee/attachment.html From anthony.minessale at gmail.com Sat May 25 01:02:06 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 May 2013 16:02:06 -0500 Subject: [Freeswitch-users] Strange issue with late negotiation In-Reply-To: References: Message-ID: A manual description of sip traffic with no trace or anything else is only going to get an educated guess. Beating the box so badly that its dropping udp packets is not going to go over well no matter how you slice it. If you have a reproducible problem, open a Jira with some traces and a more detailed description and preferably a simple reproducible test case. On Fri, May 24, 2013 at 2:27 PM, Flavio Goncalves wrote: > Hi Anthony, > > It is not a retransmission, the CSEQ is much higher and the branch is > different. It is actually another transaction. The CSEQ of the first INVITE > is 103, the second INVITE is something bigger than 400000. The biggest > issue here for us is not receiving the BYE. We had built a mechanism in the > proxy to send a BYE if not receiving the ACK in 5 seconds, but the ACK for > the REINVITE disables it. We will change the mechanism to accept only the > ACK for the first transaction, but anyway, I believe FS should have sent > the BYE. If this can be fixed (if actually is broken), it would help to > eliminate many billing issues when late negotiation is involved and the > client stops answering (caused by network failures or client overload). > > Best regards, > > > Flavio E. Goncalves > CTO - SipPulse Routing and Billing Solutions for SIP > Phone: +55 48-3025-8590, +1 248-688-0960 > VoIP: SIP:flavio at opensips.org,Skype:flaviogoncalves1 > Linkedin: www.linkedin.com/in/flavioegoncalves > Twitter: www.twitter.com/asteriskguide > > > > 2013/5/24 Anthony Minessale > >> Its probably not a re-invite but a retransmission of the same one because >> it never got the response. >> >> That RFC is trying to say you can't send another invite while you are >> waiting for a reply to the current one but you can send other things like >> info. The problem is when one side sends invite and the other end replies >> but the sender does not get that reply, they will keep sending the invite >> until it gives up. >> >> >> On Thu, May 23, 2013 at 3:38 PM, Flavio Goncalves wrote: >> >>> Hello, >>> >>> I'm having a really strange issue on billing related to late >>> negotiation. The call end up hanged with no BYE. >>> >>> There is a dialer sending thousands of calls through and OpenSIPS to a >>> termination based on FreeSwitch. I have a SIP call flow as below. Some >>> times during the day, when the volume is high, the UAC drops some 200Ok >>> and FS send a reinvite in the middle of the Initial transaction. >>> >>> ----INVITE --------------- Proxy --INVITE ---------------> FS >>> <----200 OK -------------- Proxy <---200 OK- ------------- FS >>> <--REINVITE--------------- Proxy <-REINVITE--------------- FS >>> ---481 leg does not exit-> Proxy ---481------------------> FS >>> <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS >>> ----CANCEL ---------------> Proxy ---CANCEL --------------> FS >>> <---200 Ok ---------------- Proxy <----200 Ok ------------- FS >>> >>> FS is sending a reinvite before ACK comes from the client. >>> >>> I have two questions: >>> >>> 1) Is it valid to send a reinvite in the middle of an existing >>> transaction? >>> >>> According to the RFC3261 Section 14. >>> >>> Note that a UAC MUST NOT initiate a new INVITE transaction within a >>> dialog while another INVITE transaction is in progress in either >>> direction. >>> >>> 1. If there is an ongoing INVITE client transaction, the TU MUST >>> wait until the transaction reaches the completed or terminated >>> state before initiating the new INVITE. >>> >>> 2. If there is an ongoing INVITE server transaction, the TU MUST >>> wait until the transaction reaches the confirmed or terminated >>> state before initiating the new INVITE. >>> >>> >>> But it is confusing because just below it says the opposite. >>> >>> However, a UA MAY initiate a regular transaction while an INVITE >>> transaction is in progress. A UA MAY also initiate an INVITE >>> transaction while a regular transaction is in progress. >>> >>> >>> 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the >>> ACK? >>> >>> If a UAS generates a 2xx response and never receives an ACK, it >>> SHOULD generate a BYE to terminate the dialog. >>> >>> >>> Best regards, >>> >>> Flavio E. Goncalves >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/f07d4421/attachment.html From anthony.minessale at gmail.com Sat May 25 01:28:07 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 24 May 2013 16:28:07 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: It will only work running as root I believe because it needs high privs to do realtime. If you do use centOS make sure its the latest rev of cent6, we have some horrible performance problems on the earlier revs and I don't know if they were resolved. Just make sure your kernels are as new as you can get them on the real host to avoid years of kernel performance bugs and that it has support for timerfd otherwise your VE would compile with timerfd support but not have actual access to the real syscalls for it in the host. That's why we try not to recommend virtual stuff in general as it takes some very careful setups and its hard to support from our standpoint when people run into issues. On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: > Hello, > > Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on > host and Ubuntu 10.04 LTS in VE. > Do you know maybe how to allow realtime priority in the VE for FS? Running > FS with -rp does not set the scheduler. > strace says, sched_setscheduler operation permitted, so SCHED_FF is not > set. Tried to run as root and/or use ulimit -r option, but cannot run FS > with tuned priorities.We guess, some thing missing in the host/VE > configuraton. > Any idea? > > Br, > Tamas > > > On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: > >> Im using 2.6.32 on all my boxes ... >> >> One thing that has me thinking, are there any tweaks to get MSI-X >> working best it can ? ( with proxmox ) >> there seems to be a strong bias towards one CPU for all interrupts. >> >> I could be wrong, but its something I think ive seen, and didnt see any >> clear suggestions on. >> >> Jay >> >> >> >> >> >> On 23 May 2013 01:12, Anthony Minessale wrote: >> >>> 2.6.25 or newer to get timerfd support. >>> >>> >>> >>> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >>> >>>> On 5/22/13, Anthony Minessale wrote: >>>> > You should consider centos6 or debian stable. Make sure the host >>>> kernel is >>>> > very new to get maximum results. >>>> >>>> Tony, do you mean "very new kernel" means 3.2.xx kernel? >>>> >>>> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >>>> possible. And that is what CentOS6 offers, too. >>>> >>>> However, I installed FS as openvz guest, it works fine for outgoing, >>>> but not DNAT works for incoming connections even after throroughly >>>> following >>>> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >>>> >>>> Just my two cents. >>>> >>>> >>>> >>>> > >>>> > >>>> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >>>> wrote: >>>> > >>>> >> Hello, >>>> >> >>>> >> Do you have any recommendations regarding how to set up correctly >>>> (for >>>> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >>>> >> tuneup >>>> >> the system to be rock solid? >>>> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container >>>> with FS >>>> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess >>>> that >>>> >> something is not well configured around timers, however >>>> mod_posix_timer >>>> >> did >>>> >> not help anything (running FS with -rp). We use our own bare metal >>>> and >>>> >> can >>>> >> reproduce those delatas eirher when only one VE is on the HW. >>>> >> Maybe time to check out centos6 with openvz? >>>> >> >>>> >> Any idea, recommendation, experience can be very helpful. >>>> >> >>>> >> Regards, >>>> >> Jalsot >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.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 >>>> >> >>>> >> >>>> > >>>> > >>>> > -- >>>> > Anthony Minessale II >>>> > >>>> > FreeSWITCH http://www.freeswitch.org/ >>>> > ClueCon http://www.cluecon.com/ >>>> > Twitter: http://twitter.com/FreeSWITCH_wire >>>> > >>>> > AIM: anthm >>>> > MSN:anthony_minessale at hotmail.com >>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> > IRC: irc.freenode.net #freeswitch >>>> > >>>> > FreeSWITCH Developer Conference >>>> > sip:888 at conference.freeswitch.org >>>> > googletalk:conf+888 at conference.freeswitch.org >>>> > pstn:+19193869900 >>>> > >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/23712b6c/attachment-0001.html From lloyd.aloysius at gmail.com Sat May 25 04:14:01 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Fri, 24 May 2013 20:14:01 -0400 Subject: [Freeswitch-users] dbh:query - insert id Message-ID: Hello All How to get the id value after insert a record a record using dbh:query *table_a - columns*. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130524/ff17fdd7/attachment.html From xiaofengcanyuexp at 163.com Sat May 25 05:18:16 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Sat, 25 May 2013 09:18:16 +0800 Subject: [Freeswitch-users] How to attche custom "content-type" in Freeswitch SIP message Message-ID: <201305250918139069169@163.com> Dear support, I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example. It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp". Now I can see the application/sdp is encapsulated via variable "switch_r_sdp". Is there anyway to encapsulated other customized "content-type"? Appreciate to get your reply. --------------------------------------------------------------------------------------------------------------- Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup". INVITE sip: 87896677 at dance.com;user=phone;SIP/2.0 From: "Caller" To: ;user=phone Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP at skynetwork.com Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR MIME-Version: 1.0 Content-Length: 433 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/sdp User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487 Content-Length: 142 v=0 o=- 1706944438 1706944438 IN IP4 192.168.1.105 s=ENSResip t=0 0 m=audio 6793 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --QRLVLNKeKxWDHAuwlEkR Content-Type: application/isup; version=ansi;base=ansi00 Content-Disposition: signal; handling=optional 01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 --QRLVLNKeKxWDHAuwlEkR-- Thanks ------------------- 2013-05-25 From xiaofengcanyuexp at 163.com Sat May 25 06:22:13 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Sat, 25 May 2013 10:22:13 +0800 Subject: [Freeswitch-users] How fs handle the sip-T message? Message-ID: <201305251022115628617@163.com> Hi, support team I know FS can pass out SS7 info via properity header. While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it? Thanks Windy ------------------- 2013-05-25 From dujinfang at gmail.com Sat May 25 06:33:36 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 25 May 2013 10:33:36 +0800 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: References: Message-ID: <1F177F3A96B54D738071A078F0B60576@gmail.com> Maybe try the RETURNING clause ? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: > Hello All > > How to get the id value after insert a record a record using dbh:query > > table_a - columns. > > id - auto increment > field1 > field2 > > > dbh:query("insert into table_a ( field1,field2) values ('11','Test')") > > > After insert how to get the table_a - id value for the inserted record? > > Thanks > Lloyd > > > > > _________________________________________________________________________ > 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://wiki.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/20130525/a192c916/attachment.html From dujinfang at gmail.com Sat May 25 06:39:03 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 25 May 2013 10:39:03 +0800 Subject: [Freeswitch-users] How to attche custom "content-type" in Freeswitch SIP message In-Reply-To: <201305250918139069169@163.com> References: <201305250918139069169@163.com> Message-ID: <50D08AA095C54A7188B2C7E9F833D199@gmail.com> FS defined SOFIA_MULTIPART_PREFIX which should can support multipart body. not sure if you can find the wiki. maybe you should look the source code and update the wiki when you find out. I had tried that but I forget the procedure. it should be some chan vars like sip_mp_ ... -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, May 25, 2013 at 9:18 AM, xiaofengcanyuexp at 163.com wrote: > Dear support, > > I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example. > > It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp". > Now I can see the application/sdp is encapsulated via variable "switch_r_sdp". > Is there anyway to encapsulated other customized "content-type"? > > Appreciate to get your reply. > --------------------------------------------------------------------------------------------------------------- > Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup". > > INVITE sip: 87896677 at dance.com (mailto:87896677 at dance.com);user=phone;SIP/2.0 > From: "Caller" > To: ;user=phone > Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP at skynetwork.com (mailto:QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP at skynetwork.com) > Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR > MIME-Version: 1.0 > Content-Length: 433 > --QRLVLNKeKxWDHAuwlEkR > Content-Type: application/sdp > User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487 > Content-Length: 142 > > v=0 > o=- 1706944438 1706944438 IN IP4 192.168.1.105 > s=ENSResip > t=0 0 > m=audio 6793 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > --QRLVLNKeKxWDHAuwlEkR > Content-Type: application/isup; version=ansi;base=ansi00 > Content-Disposition: signal; handling=optional > > 01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 > --QRLVLNKeKxWDHAuwlEkR-- > > > Thanks > ------------------- > 2013-05-25 > > > > > _________________________________________________________________________ > 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://wiki.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/20130525/94ef1ef5/attachment.html From dujinfang at gmail.com Sat May 25 06:46:56 2013 From: dujinfang at gmail.com (Seven Du) Date: Sat, 25 May 2013 10:46:56 +0800 Subject: [Freeswitch-users] How fs handle the sip-T message? In-Reply-To: <201305251022115628617@163.com> References: <201305251022115628617@163.com> Message-ID: <18F391469E044E38B465F4CF53E14823@gmail.com> Bottom line is you should try, by looking the log and possibly using the "info" app you should can find out. I bet you should be receive the SIP-T anyway. I remembered some discussion about this, maybe you can google against this list and find out more. I think Sangoma has some private implementation which can help SS7 passthru over SIP. Or maybe try to put the SIP-T int the new b64 codec? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, May 25, 2013 at 10:22 AM, xiaofengcanyuexp at 163.com wrote: > Hi, support team > > I know FS can pass out SS7 info via properity header. > While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it? > > > Thanks > Windy > ------------------- > 2013-05-25 > > > > > _________________________________________________________________________ > 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://wiki.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/20130525/eefab9d0/attachment-0001.html From bdfoster at davri.com Sat May 25 06:55:45 2013 From: bdfoster at davri.com (Brian Foster) Date: Fri, 24 May 2013 22:55:45 -0400 Subject: [Freeswitch-users] How fs handle the sip-T message? In-Reply-To: <18F391469E044E38B465F4CF53E14823@gmail.com> References: <201305251022115628617@163.com> <18F391469E044E38B465F4CF53E14823@gmail.com> Message-ID: This? http://www.radisys.com/products/trillium/ss7-sigtran/ - BDF On May 24, 2013 10:53 PM, "Seven Du" wrote: > Bottom line is you should try, by looking the log and possibly using the > "info" app you should can find out. I bet you should be receive the SIP-T > anyway. > > I remembered some discussion about this, maybe you can google against this > list and find out more. I think Sangoma has some private implementation > which can help SS7 passthru over SIP. Or maybe try to put the SIP-T int the > new b64 codec? > > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Saturday, May 25, 2013 at 10:22 AM, xiaofengcanyuexp at 163.com wrote: > > Hi, support team > > I know FS can pass out SS7 info via properity header. > While, how it handles an incoming SIP-T message with SS7 info encapsulated > in "content-type". Will it parse it or discard it? > > Thanks > Windy > ------------------- > 2013-05-25 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130524/595a5da3/attachment.html From j_mj at aol.com Sat May 25 07:59:54 2013 From: j_mj at aol.com (John M) Date: Fri, 24 May 2013 23:59:54 -0400 (EDT) Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <1F177F3A96B54D738071A078F0B60576@gmail.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> Message-ID: <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Hi Seven Du, I'd really like to know if this is possible too, couldn't find it documented anywhere. Instead of being cryptic, if you know the answer won't you please help by explaining what the RETURNING clause is and how to use it? Does it somehow return mysql_insert_id()? How should we use it. You help is invaluable and is contributing to the freeswitch community. -Jm -----Original Message----- From: Seven Du To: FreeSWITCH Users Help Sent: Sat, May 25, 2013 12:52 pm Subject: Re: [Freeswitch-users] dbh:query - insert id Maybe try the RETURNING clause ? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: Hello All How to get the id value after insert a record a record using dbh:query table_a - columns. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130524/7ba0e708/attachment.html From jalsot at gmail.com Sat May 25 09:52:28 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Sat, 25 May 2013 07:52:28 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: We've tried as root user but as I remember it was not able to set up the priority but will recheck to be sure. Another interesting thing is that when I tried to set a running process scheduler with chrt, got operation not permitted (as root of course), so I guess, something has to be tuned in the VE or on the host. We will try on bare metal centos6+ovz on the host. We try the latest centos6 with latest openvz kernel (due to security requirements we run on latest stable kernel and OS versions). Could you give some info about those horrible performance problems to let us check whether it still applies or not? (we've found only mysql create table performance degradation due to ext4 - where the solution could be barrier=0 yet, no other problems). Actually, how can I be sure that timerfd is used? strace? I'm nearly sure that timerfd works fine in FS. Yep, it would be much simpler without virtualization, and much harder from another perspective. Probably lxc, kvm and xen aren't much better regarding realtime stuff... Regards, Tamas On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > It will only work running as root I believe because it needs high privs to > do realtime. > If you do use centOS make sure its the latest rev of cent6, we have some > horrible performance problems on the earlier revs and I don't know if they > were resolved. > Just make sure your kernels are as new as you can get them on the real > host to avoid years of kernel performance bugs and that it has support for > timerfd otherwise your VE would compile with timerfd support but not have > actual access to the real syscalls for it in the host. > > That's why we try not to recommend virtual stuff in general as it takes > some very careful setups and its hard to support from our standpoint when > people run into issues. > > > > > > On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: > >> Hello, >> >> Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on >> host and Ubuntu 10.04 LTS in VE. >> Do you know maybe how to allow realtime priority in the VE for FS? >> Running FS with -rp does not set the scheduler. >> strace says, sched_setscheduler operation permitted, so SCHED_FF is not >> set. Tried to run as root and/or use ulimit -r option, but cannot run FS >> with tuned priorities.We guess, some thing missing in the host/VE >> configuraton. >> Any idea? >> >> Br, >> Tamas >> >> >> On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: >> >>> Im using 2.6.32 on all my boxes ... >>> >>> One thing that has me thinking, are there any tweaks to get MSI-X >>> working best it can ? ( with proxmox ) >>> there seems to be a strong bias towards one CPU for all interrupts. >>> >>> I could be wrong, but its something I think ive seen, and didnt see any >>> clear suggestions on. >>> >>> Jay >>> >>> >>> >>> >>> >>> On 23 May 2013 01:12, Anthony Minessale wrote: >>> >>>> 2.6.25 or newer to get timerfd support. >>>> >>>> >>>> >>>> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >>>> >>>>> On 5/22/13, Anthony Minessale wrote: >>>>> > You should consider centos6 or debian stable. Make sure the host >>>>> kernel is >>>>> > very new to get maximum results. >>>>> >>>>> Tony, do you mean "very new kernel" means 3.2.xx kernel? >>>>> >>>>> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >>>>> possible. And that is what CentOS6 offers, too. >>>>> >>>>> However, I installed FS as openvz guest, it works fine for outgoing, >>>>> but not DNAT works for incoming connections even after throroughly >>>>> following >>>>> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >>>>> >>>>> Just my two cents. >>>>> >>>>> >>>>> >>>>> > >>>>> > >>>>> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >>>>> wrote: >>>>> > >>>>> >> Hello, >>>>> >> >>>>> >> Do you have any recommendations regarding how to set up correctly >>>>> (for >>>>> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >>>>> >> tuneup >>>>> >> the system to be rock solid? >>>>> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container >>>>> with FS >>>>> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess >>>>> that >>>>> >> something is not well configured around timers, however >>>>> mod_posix_timer >>>>> >> did >>>>> >> not help anything (running FS with -rp). We use our own bare metal >>>>> and >>>>> >> can >>>>> >> reproduce those delatas eirher when only one VE is on the HW. >>>>> >> Maybe time to check out centos6 with openvz? >>>>> >> >>>>> >> Any idea, recommendation, experience can be very helpful. >>>>> >> >>>>> >> Regards, >>>>> >> Jalsot >>>>> >> >>>>> >> >>>>> _________________________________________________________________________ >>>>> >> Professional FreeSWITCH Consulting Services: >>>>> >> consulting at freeswitch.org >>>>> >> http://www.freeswitchsolutions.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 >>>>> >> >>>>> >> >>>>> > >>>>> > >>>>> > -- >>>>> > Anthony Minessale II >>>>> > >>>>> > FreeSWITCH http://www.freeswitch.org/ >>>>> > ClueCon http://www.cluecon.com/ >>>>> > Twitter: http://twitter.com/FreeSWITCH_wire >>>>> > >>>>> > AIM: anthm >>>>> > MSN:anthony_minessale at hotmail.com >>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> > IRC: irc.freenode.net #freeswitch >>>>> > >>>>> > FreeSWITCH Developer Conference >>>>> > sip:888 at conference.freeswitch.org >>>>> > googletalk:conf+888 at conference.freeswitch.org >>>>> > pstn:+19193869900 >>>>> > >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130525/c315ee37/attachment-0001.html From xiaofengcanyuexp at 163.com Sat May 25 11:10:21 2013 From: xiaofengcanyuexp at 163.com (=?utf-8?B?eGlhb2ZlbmdjYW55dWV4cEAxNjMuY29t?=) Date: Sat, 25 May 2013 15:10:21 +0800 Subject: [Freeswitch-users] =?utf-8?q?How_fs_handle_the_sip-T_message=3F?= References: <201305251022115628617@163.com>, <18F391469E044E38B465F4CF53E14823@gmail.com> Message-ID: <201305251510175625437@163.com> Seven, Thanks. So far, sangoma passthought the SS7 by extracting and copying the properity to freetdm channel info. It utilizes some structure siConEvnt(IAM,ANM) and siCnStEvnt(ACM...). Here now one INVITE comes into FS, but the "application/isup" in the SIP message dispears in the FS outbound. I'll do more debug and google. Thank you so much. Windy ?? ======== 2013-05-25 10:50:05 Original Message? ======== Bottom line is you should try, by looking the log and possibly using the "info" app you should can find out. I bet you should be receive the SIP-T anyway. I remembered some discussion about this, maybe you can google against this list and find out more. I think Sangoma has some private implementation which can help SS7 passthru over SIP. Or maybe try to put the SIP-T int the new b64 codec? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Saturday, May 25, 2013 at 10:22 AM, xiaofengcanyuexp at 163.com wrote: Hi, support team I know FS can pass out SS7 info via properity header. While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it? Thanks Windy ------------------- 2013-05-25 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 = = = = = = = = = = = = = = = = = = = = = = Thanks Windy ?????????????? ?????????????? ??????????????? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130525/542c0bdc/attachment.html From sertys at gmail.com Sat May 25 11:47:34 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sat, 25 May 2013 09:47:34 +0200 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Message-ID: It is true that the luasql driver is overly basic and poorly documented . Unfortunately mysql doesn't support RETURNING clause like pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); right after the insert query. I cannot guarantee it works due to the unknown nature(to me that is) of the luasql transaction handling, but it should keep a transaction open for as long as a db handler lives. On May 25, 2013 7:03 AM, "John M" wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it > documented anywhere. > > Instead of being cryptic, if you know the answer won't you please help by > explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > Maybe try the RETURNING clause ? > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: > > Hello All > > How to get the id value after insert a record a record using dbh:query > > *table_a - columns*. > > id - auto increment > field1 > field2 > > > dbh:query("insert into table_a ( field1,field2) values ('11','Test')") > > > After insert how to get the table_a - id value for the inserted record? > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130525/acd98346/attachment.html From j_mj at aol.com Sat May 25 12:01:30 2013 From: j_mj at aol.com (John M) Date: Sat, 25 May 2013 04:01:30 -0400 (EDT) Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Message-ID: <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> Hi Daniel, Thanks for your description, it is much appreciated. :-) 5 word one liners from people too lazy to explain properly would really be best if they didn't reply at all. Cheers, thanks again. -Jm -----Original Message----- From: Daniel Ivanov To: FreeSWITCH Users Help Sent: Sat, May 25, 2013 5:57 pm Subject: Re: [Freeswitch-users] dbh:query - insert id It is true that the luasql driver is overly basic and poorly documented . Unfortunately mysql doesn't support RETURNING clause like pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); right after the insert query. I cannot guarantee it works due to the unknown nature(to me that is) of the luasql transaction handling, but it should keep a transaction open for as long as a db handler lives. On May 25, 2013 7:03 AM, "John M" wrote: Hi Seven Du, I'd really like to know if this is possible too, couldn't find it documented anywhere. Instead of being cryptic, if you know the answer won't you please help by explaining what the RETURNING clause is and how to use it? Does it somehow return mysql_insert_id()? How should we use it. You help is invaluable and is contributing to the freeswitch community. -Jm -----Original Message----- From: Seven Du To: FreeSWITCH Users Help Sent: Sat, May 25, 2013 12:52 pm Subject: Re: [Freeswitch-users] dbh:query - insert id Maybe try the RETURNING clause ? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: Hello All How to get the id value after insert a record a record using dbh:query table_a - columns. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130525/d1221545/attachment-0001.html From nasida at live.ru Sat May 25 13:52:56 2013 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 25 May 2013 13:52:56 +0400 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: , , , , , , , Message-ID: Tamas, I think you can find info about performance problems with CentOS 6 on jira. For example. http://jira.freeswitch.org/browse/FS-4291 We also wanted to use timerfd (without virtualization) and made a moving to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen sometimes. As a result we had to move on centos 5.9 again. I would like to join issue. how can I be sure that timerfd is used? Regards, Yuriy Date: Sat, 25 May 2013 07:52:28 +0200 From: jalsot at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OpenVZ tuning tips We've tried as root user but as I remember it was not able to set up the priority but will recheck to be sure. Another interesting thing is that when I tried to set a running process scheduler with chrt, got operation not permitted (as root of course), so I guess, something has to be tuned in the VE or on the host. We will try on bare metal centos6+ovz on the host. We try the latest centos6 with latest openvz kernel (due to security requirements we run on latest stable kernel and OS versions). Could you give some info about those horrible performance problems to let us check whether it still applies or not? (we've found only mysql create table performance degradation due to ext4 - where the solution could be barrier=0 yet, no other problems). Actually, how can I be sure that timerfd is used? strace? I'm nearly sure that timerfd works fine in FS. Yep, it would be much simpler without virtualization, and much harder from another perspective. Probably lxc, kvm and xen aren't much better regarding realtime stuff... Regards, Tamas On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale wrote: It will only work running as root I believe because it needs high privs to do realtime.If you do use centOS make sure its the latest rev of cent6, we have some horrible performance problems on the earlier revs and I don't know if they were resolved. Just make sure your kernels are as new as you can get them on the real host to avoid years of kernel performance bugs and that it has support for timerfd otherwise your VE would compile with timerfd support but not have actual access to the real syscalls for it in the host. That's why we try not to recommend virtual stuff in general as it takes some very careful setups and its hard to support from our standpoint when people run into issues. On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: Hello, Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on host and Ubuntu 10.04 LTS in VE. Do you know maybe how to allow realtime priority in the VE for FS? Running FS with -rp does not set the scheduler. strace says, sched_setscheduler operation permitted, so SCHED_FF is not set. Tried to run as root and/or use ulimit -r option, but cannot run FS with tuned priorities.We guess, some thing missing in the host/VE configuraton. Any idea? Br, Tamas On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: Im using 2.6.32 on all my boxes ... One thing that has me thinking, are there any tweaks to get MSI-X working best it can ? ( with proxmox ) there seems to be a strong bias towards one CPU for all interrupts. I could be wrong, but its something I think ive seen, and didnt see any clear suggestions on. Jay On 23 May 2013 01:12, Anthony Minessale wrote: 2.6.25 or newer to get timerfd support. On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: On 5/22/13, Anthony Minessale wrote: > You should consider centos6 or debian stable. Make sure the host kernel is > very new to get maximum results. Tony, do you mean "very new kernel" means 3.2.xx kernel? Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not possible. And that is what CentOS6 offers, too. However, I installed FS as openvz guest, it works fine for outgoing, but not DNAT works for incoming connections even after throroughly following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. Just my two cents. > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky wrote: > >> Hello, >> >> Do you have any recommendations regarding how to set up correctly (for >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> tuneup >> the system to be rock solid? >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> something is not well configured around timers, however mod_posix_timer >> did >> not help anything (running FS with -rp). We use our own bare metal and >> can >> reproduce those delatas eirher when only one VE is on the HW. >> Maybe time to check out centos6 with openvz? >> >> Any idea, recommendation, experience can be very helpful. >> >> Regards, >> Jalsot >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130525/a60e13e5/attachment-0001.html From nasida at live.ru Sat May 25 14:02:05 2013 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 25 May 2013 14:02:05 +0400 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: , , , , , , , , Message-ID: I mean to join question (sure not issue) :) From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Subject: RE: [Freeswitch-users] OpenVZ tuning tips Date: Sat, 25 May 2013 13:52:56 +0400 Tamas, I think you can find info about performance problems with CentOS 6 on jira. For example. http://jira.freeswitch.org/browse/FS-4291 We also wanted to use timerfd (without virtualization) and made a moving to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen sometimes. As a result we had to move on centos 5.9 again. I would like to join issue. how can I be sure that timerfd is used? Regards, Yuriy Date: Sat, 25 May 2013 07:52:28 +0200 From: jalsot at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] OpenVZ tuning tips We've tried as root user but as I remember it was not able to set up the priority but will recheck to be sure. Another interesting thing is that when I tried to set a running process scheduler with chrt, got operation not permitted (as root of course), so I guess, something has to be tuned in the VE or on the host. We will try on bare metal centos6+ovz on the host. We try the latest centos6 with latest openvz kernel (due to security requirements we run on latest stable kernel and OS versions). Could you give some info about those horrible performance problems to let us check whether it still applies or not? (we've found only mysql create table performance degradation due to ext4 - where the solution could be barrier=0 yet, no other problems). Actually, how can I be sure that timerfd is used? strace? I'm nearly sure that timerfd works fine in FS. Yep, it would be much simpler without virtualization, and much harder from another perspective. Probably lxc, kvm and xen aren't much better regarding realtime stuff... Regards, Tamas On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale wrote: It will only work running as root I believe because it needs high privs to do realtime.If you do use centOS make sure its the latest rev of cent6, we have some horrible performance problems on the earlier revs and I don't know if they were resolved. Just make sure your kernels are as new as you can get them on the real host to avoid years of kernel performance bugs and that it has support for timerfd otherwise your VE would compile with timerfd support but not have actual access to the real syscalls for it in the host. That's why we try not to recommend virtual stuff in general as it takes some very careful setups and its hard to support from our standpoint when people run into issues. On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: Hello, Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on host and Ubuntu 10.04 LTS in VE. Do you know maybe how to allow realtime priority in the VE for FS? Running FS with -rp does not set the scheduler. strace says, sched_setscheduler operation permitted, so SCHED_FF is not set. Tried to run as root and/or use ulimit -r option, but cannot run FS with tuned priorities.We guess, some thing missing in the host/VE configuraton. Any idea? Br, Tamas On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: Im using 2.6.32 on all my boxes ... One thing that has me thinking, are there any tweaks to get MSI-X working best it can ? ( with proxmox ) there seems to be a strong bias towards one CPU for all interrupts. I could be wrong, but its something I think ive seen, and didnt see any clear suggestions on. Jay On 23 May 2013 01:12, Anthony Minessale wrote: 2.6.25 or newer to get timerfd support. On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: On 5/22/13, Anthony Minessale wrote: > You should consider centos6 or debian stable. Make sure the host kernel is > very new to get maximum results. Tony, do you mean "very new kernel" means 3.2.xx kernel? Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not possible. And that is what CentOS6 offers, too. However, I installed FS as openvz guest, it works fine for outgoing, but not DNAT works for incoming connections even after throroughly following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. Just my two cents. > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky wrote: > >> Hello, >> >> Do you have any recommendations regarding how to set up correctly (for >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> tuneup >> the system to be rock solid? >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> something is not well configured around timers, however mod_posix_timer >> did >> not help anything (running FS with -rp). We use our own bare metal and >> can >> reproduce those delatas eirher when only one VE is on the HW. >> Maybe time to check out centos6 with openvz? >> >> Any idea, recommendation, experience can be very helpful. >> >> Regards, >> Jalsot >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130525/16f34ebc/attachment-0001.html From flavio at sippulse.com Sat May 25 15:01:18 2013 From: flavio at sippulse.com (Flavio Goncalves) Date: Sat, 25 May 2013 08:01:18 -0300 Subject: [Freeswitch-users] Strange issue with late negotiation In-Reply-To: References: Message-ID: Hi Anthony, Thanks for your time, I will open the case with the traces. It is reproducible and occurs once or twice per week, but to reproduce in the LAB you would need to have sipp with the scenario, it won't be reproducible if the client is not dropping the 200Ok. I'm managing now to drop the RE-INVITES if the initial ACK is not received. This will give time for the first transaction to complete or simply timeout. I'll be testing during this week. Best regards, Flavio E. Goncalves CTO - SipPulse Routing and Billing Solutions for SIP Phone: +55 48-3025-8590, +1 248-688-0960 VoIP: SIP:flavio at opensips.org,Skype:flaviogoncalves1 Linkedin: www.linkedin.com/in/flavioegoncalves Twitter: www.twitter.com/asteriskguide 2013/5/24 Anthony Minessale > A manual description of sip traffic with no trace or anything else is only > going to get an educated guess. > Beating the box so badly that its dropping udp packets is not going to go > over well no matter how you slice it. > If you have a reproducible problem, open a Jira with some traces and a > more detailed description and preferably a simple reproducible test case. > > > > > On Fri, May 24, 2013 at 2:27 PM, Flavio Goncalves wrote: > >> Hi Anthony, >> >> It is not a retransmission, the CSEQ is much higher and the branch is >> different. It is actually another transaction. The CSEQ of the first INVITE >> is 103, the second INVITE is something bigger than 400000. The biggest >> issue here for us is not receiving the BYE. We had built a mechanism in the >> proxy to send a BYE if not receiving the ACK in 5 seconds, but the ACK for >> the REINVITE disables it. We will change the mechanism to accept only the >> ACK for the first transaction, but anyway, I believe FS should have sent >> the BYE. If this can be fixed (if actually is broken), it would help to >> eliminate many billing issues when late negotiation is involved and the >> client stops answering (caused by network failures or client overload). >> >> Best regards, >> >> >> Flavio E. Goncalves >> CTO - SipPulse Routing and Billing Solutions for SIP >> Phone: +55 48-3025-8590, +1 248-688-0960 >> VoIP: SIP:flavio at opensips.org,Skype:flaviogoncalves1 >> Linkedin: www.linkedin.com/in/flavioegoncalves >> Twitter: www.twitter.com/asteriskguide >> >> >> >> 2013/5/24 Anthony Minessale >> >>> Its probably not a re-invite but a retransmission of the same one >>> because it never got the response. >>> >>> That RFC is trying to say you can't send another invite while you are >>> waiting for a reply to the current one but you can send other things like >>> info. The problem is when one side sends invite and the other end replies >>> but the sender does not get that reply, they will keep sending the invite >>> until it gives up. >>> >>> >>> On Thu, May 23, 2013 at 3:38 PM, Flavio Goncalves >> > wrote: >>> >>>> Hello, >>>> >>>> I'm having a really strange issue on billing related to late >>>> negotiation. The call end up hanged with no BYE. >>>> >>>> There is a dialer sending thousands of calls through and OpenSIPS to a >>>> termination based on FreeSwitch. I have a SIP call flow as below. Some >>>> times during the day, when the volume is high, the UAC drops some 200Ok >>>> and FS send a reinvite in the middle of the Initial transaction. >>>> >>>> ----INVITE --------------- Proxy --INVITE ---------------> FS >>>> <----200 OK -------------- Proxy <---200 OK- ------------- FS >>>> <--REINVITE--------------- Proxy <-REINVITE--------------- FS >>>> ---481 leg does not exit-> Proxy ---481------------------> FS >>>> <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS >>>> ----CANCEL ---------------> Proxy ---CANCEL --------------> FS >>>> <---200 Ok ---------------- Proxy <----200 Ok ------------- FS >>>> >>>> FS is sending a reinvite before ACK comes from the client. >>>> >>>> I have two questions: >>>> >>>> 1) Is it valid to send a reinvite in the middle of an existing >>>> transaction? >>>> >>>> According to the RFC3261 Section 14. >>>> >>>> Note that a UAC MUST NOT initiate a new INVITE transaction within a >>>> dialog while another INVITE transaction is in progress in either >>>> direction. >>>> >>>> 1. If there is an ongoing INVITE client transaction, the TU MUST >>>> wait until the transaction reaches the completed or terminated >>>> state before initiating the new INVITE. >>>> >>>> 2. If there is an ongoing INVITE server transaction, the TU MUST >>>> wait until the transaction reaches the confirmed or terminated >>>> state before initiating the new INVITE. >>>> >>>> >>>> But it is confusing because just below it says the opposite. >>>> >>>> However, a UA MAY initiate a regular transaction while an INVITE >>>> transaction is in progress. A UA MAY also initiate an INVITE >>>> transaction while a regular transaction is in progress. >>>> >>>> >>>> 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the >>>> ACK? >>>> >>>> If a UAS generates a 2xx response and never receives an ACK, it >>>> SHOULD generate a BYE to terminate the dialog. >>>> >>>> >>>> Best regards, >>>> >>>> Flavio E. Goncalves >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130525/91569a3d/attachment.html From freeswitch-list at puzzled.xs4all.nl Sat May 25 16:04:47 2013 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 25 May 2013 14:04:47 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: , , , , , , , Message-ID: <51A0A8DF.4070609@puzzled.xs4all.nl> On 05/25/2013 11:52 AM, Yuriy Nasida wrote: > Tamas, > > I think you can find info about performance problems with CentOS 6 on jira. > For example. http://jira.freeswitch.org/browse/FS-4291 > > We also wanted to use timerfd (without virtualization) and made a > moving to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got > frozen sometimes. As a result we had to move on centos 5.9 again. For starters I would update to CentOS 6.4 and see if the problem persists. The CentOS 6.2/6.3 kernels contain a local root exploit which is fixed in the latest kernel update. So updating to 6.4 + updates is recommended. Regards, Patrick From william.king at quentustech.com Sat May 25 17:19:15 2013 From: william.king at quentustech.com (William King) Date: Sat, 25 May 2013 06:19:15 -0700 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Message-ID: <51A0BA53.4030007@quentustech.com> http://lmgtfy.com/?q=sql+returning+clause http://en.wikipedia.org/wiki/Insert_%28SQL%29#Retrieving_the_key http://www.postgresql.org/docs/9.2/static/sql-insert.html Much less useful unless you have no other choice than to use mysql: http://dev.mysql.com/doc/refman/5.0/en/information-functions.html#function_last-insert-id 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 05/24/2013 08:59 PM, John M wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it > documented anywhere. > > Instead of being cryptic, if you know the answer won't you please help > by explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > Maybe try the RETURNING clause ? > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: >> Hello All >> >> How to get the id value after insert a record a record using dbh:query >> >> _table_a - columns_. >> >> id - auto increment >> field1 >> field2 >> >> >> dbh:query("insert into table_a ( field1,field2) values ('11','Test')") >> >> >> After insert how to get the table_a - id value for the inserted record? >> >> Thanks >> Lloyd >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From j_mj at aol.com Sat May 25 18:23:27 2013 From: j_mj at aol.com (John M) Date: Sat, 25 May 2013 10:23:27 -0400 (EDT) Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <51A0BA53.4030007@quentustech.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <51A0BA53.4030007@quentustech.com> Message-ID: <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> Yep, super helpful.. /sarcasm Point us to general sql related pages, none of which are related to freeswitch and the way the language interacts with the db .. We all know that insert_id exists that's why we are asking. What we don't know is how specifically the freeswitch:dbh handler interacts and can return the inserted record id. -----Original Message----- From: William King To: freeswitch-users Sent: Sat, May 25, 2013 11:24 pm Subject: Re: [Freeswitch-users] dbh:query - insert id http://lmgtfy.com/?q=sql+returning+clause http://en.wikipedia.org/wiki/Insert_%28SQL%29#Retrieving_the_key http://www.postgresql.org/docs/9.2/static/sql-insert.html Much less useful unless you have no other choice than to use mysql: http://dev.mysql.com/doc/refman/5.0/en/information-functions.html#function_last-insert-id 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 05/24/2013 08:59 PM, John M wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it > documented anywhere. > > Instead of being cryptic, if you know the answer won't you please help > by explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > Maybe try the RETURNING clause ? > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: >> Hello All >> >> How to get the id value after insert a record a record using dbh:query >> >> _table_a - columns_. >> >> id - auto increment >> field1 >> field2 >> >> >> dbh:query("insert into table_a ( field1,field2) values ('11','Test')") >> >> >> After insert how to get the table_a - id value for the inserted record? >> >> Thanks >> Lloyd >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130525/5c52550a/attachment.html From krice at freeswitch.org Sat May 25 18:42:45 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 25 May 2013 09:42:45 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: Message-ID: I actually dropped using Centos6 and moved to debian to get both timerfd and to get performance at the same time. And things started working much nicer... Not sure if they even fixed the performance issues on centos tho On 5/25/13 4:52 AM, "Yuriy Nasida" wrote: > Tamas, > > I think you can find info about performance problems with CentOS 6 on jira. > For example. http://jira.freeswitch.org/browse/FS-4291 > > We also wanted to use timerfd (without virtualization) and made a moving to > latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen sometimes. > As a result we had to move on centos 5.9 again. > > I would like to join issue. how can I be sure that timerfd is used? > > Regards, > Yuriy > > > Date: Sat, 25 May 2013 07:52:28 +0200 > From: jalsot at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] OpenVZ tuning tips > > We've tried as root user but as I remember it was not able to set up the > priority but will recheck to be sure. Another interesting thing is that when I > tried to set a running process scheduler with chrt, got operation not > permitted (as root of course), so I guess, something has to be tuned in the VE > or on the host. We will try on bare metal centos6+ovz on the host. > We try the latest centos6 with latest openvz kernel (due to security > requirements we run on latest stable kernel and OS versions). > Could you give some info about those horrible performance problems to let us > check whether it still applies or not? (we've found only mysql create table > performance degradation due to ext4 - where the solution could be barrier=0 > yet, no other problems). > Actually, how can I be sure that timerfd is used? strace? I'm nearly sure that > timerfd works fine in FS. > > Yep, it would be much simpler without virtualization, and much harder from > another perspective. Probably lxc, kvm and xen aren't much better regarding > realtime stuff... > > Regards, > Tamas > > > On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale > wrote: >> It will only work running as root I believe because it needs high privs to do >> realtime. >> If you do use centOS make sure its the latest rev of cent6, we have some >> horrible performance problems on the earlier revs and I don't know if they >> were resolved. >> Just make sure your kernels are as new as you can get them on the real host >> to avoid years of kernel performance bugs and that it has support for timerfd >> otherwise your VE would compile with timerfd support but not have actual >> access to the real syscalls for it in the host. >> >> That's why we try not to recommend virtual stuff in general as it takes some >> very careful setups and its hard to support from our standpoint when people >> run into issues. >> >> >> >> >> >> On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: >>> Hello, >>> >>> Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on >>> host and Ubuntu 10.04 LTS in VE. >>> Do you know maybe how to allow realtime priority in the VE for FS? Running >>> FS with -rp does not set the scheduler. >>> strace says, sched_setscheduler operation permitted, so SCHED_FF is not set. >>> Tried to run as root and/or use ulimit -r option, but cannot run FS with >>> tuned priorities.We guess, some thing missing in the host/VE configuraton. >>> Any idea? >>> >>> Br, >>> Tamas >>> >>> >>> On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: >>>> Im using 2.6.32 on all my boxes ... >>>> >>>> One thing that has me thinking, are there any tweaks to get MSI-X working >>>> best it can ? ( with proxmox ) >>>> there seems to be a strong bias towards one CPU for all interrupts. >>>> >>>> I could be wrong, but its something I think ive seen, and didnt see any >>>> clear suggestions on. >>>> >>>> Jay >>>> >>>> >>>> >>>> >>>> >>>> On 23 May 2013 01:12, Anthony Minessale >>>> wrote: >>>>> 2.6.25 or newer to get timerfd support. >>>>> >>>>> >>>>> >>>>> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >>>>>> On 5/22/13, Anthony Minessale wrote: >>>>>>> > You should consider centos6 or debian stable. Make sure the host >>>>>>> kernel is >>>>>>> > very new to get maximum results. >>>>>> >>>>>> Tony, do you mean "very new kernel" means 3.2.xx kernel? >>>>>> >>>>>> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >>>>>> possible. And that is what CentOS6 offers, too. >>>>>> >>>>>> However, I installed FS as openvz guest, it works fine for outgoing, >>>>>> but not DNAT works for incoming connections even after throroughly >>>>>> following >>>>>> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >>>>>> >>>>>> Just my two cents. >>>>>> >>>>>> >>>>>> >>>>>>> > >>>>>>> > >>>>>>> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >>>>>>> wrote: >>>>>>> > >>>>>>>> >> Hello, >>>>>>>> >> >>>>>>>> >> Do you have any recommendations regarding how to set up correctly >>>>>>>> (for >>>>>>>> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >>>>>>>> >> tuneup >>>>>>>> >> the system to be rock solid? >>>>>>>> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container >>>>>>>> with FS >>>>>>>> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess >>>>>>>> that >>>>>>>> >> something is not well configured around timers, however >>>>>>>> mod_posix_timer >>>>>>>> >> did >>>>>>>> >> not help anything (running FS with -rp). We use our own bare metal >>>>>>>> and >>>>>>>> >> can >>>>>>>> >> reproduce those delatas eirher when only one VE is on the HW. >>>>>>>> >> Maybe time to check out centos6 with openvz? >>>>>>>> >> >>>>>>>> >> Any idea, recommendation, experience can be very helpful. >>>>>>>> >> >>>>>>>> >> Regards, >>>>>>>> >> Jalsot >>>>>>>> >> >>>>>>>> >> >>>>>>>> _________________________________________________________________________ >>>>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>>>> >> consulting at freeswitch.org >>>>>>>> >> http://www.freeswitchsolutions.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 >>>>>>>> >> >>>>>>>> >> >>>>>>> > >>>>>>> > >>>>>>> > -- >>>>>>> > Anthony Minessale II >>>>>>> > >>>>>>> > FreeSWITCH http://www.freeswitch.org/ >>>>>>> > ClueCon http://www.cluecon.com/ >>>>>>> > Twitter: http://twitter.com/FreeSWITCH_wire >>>>>>> > >>>>>>> > AIM: anthm >>>>>>> > MSN:anthony_minessale at hotmail.com >>>>>>> >>>>>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>>>> >>>>>>> > IRC: irc.freenode.net #freeswitch >>>>>>> > >>>>>>> > FreeSWITCH Developer Conference >>>>>>> > sip:888 at conference.freeswitch.org >>>>>>> >>>>>>> > googletalk:conf+888 at conference.freeswitch.org >>>>>>> >>>>>>> > pstn:+19193869900 >>>>>>> > >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130525/b3e12c8e/attachment-0001.html From lloyd.aloysius at gmail.com Sat May 25 19:01:20 2013 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Sat, 25 May 2013 11:01:20 -0400 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <51A0BA53.4030007@quentustech.com> <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> Message-ID: William, Thank you for the reply. Question was related to freeswitch dbh driver. Not related to sql specific. Most of the SQL Driver have a method to get the id after the insert record. My questions was does freeswitch dbh have any method to get the id? For Example dbh have a method dbh:affected_rows() for insert , update. Like this do we have a method for ID? Thanks Lloyd * * * * On Sat, May 25, 2013 at 10:23 AM, John M wrote: > Yep, super helpful.. /sarcasm > > Point us to general sql related pages, none of which are related to > freeswitch and the way the language interacts with the db .. > > We all know that insert_id exists that's why we are asking. What we don't > know is how specifically the freeswitch:dbh handler interacts and can > return the inserted record id. > > > -----Original Message----- > From: William King > To: freeswitch-users > Sent: Sat, May 25, 2013 11:24 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > http://lmgtfy.com/?q=sql+returning+clause > http://en.wikipedia.org/wiki/Insert_%28SQL%29#Retrieving_the_key > http://www.postgresql.org/docs/9.2/static/sql-insert.html > > Much less useful unless you have no other choice than to use mysql:http://dev.mysql.com/doc/refman/5.0/en/information-functions.html#function_last-insert-id > > 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 05/24/2013 08:59 PM, John M wrote: > > Hi Seven Du, > > > > I'd really like to know if this is possible too, couldn't find it > > documented anywhere. > > > > Instead of being cryptic, if you know the answer won't you please help > > by explaining what the RETURNING clause is and how to use it? > > > > Does it somehow return mysql_insert_id()? > > > > How should we use it. > > > > You help is invaluable and is contributing to the freeswitch community. > > > > -Jm > > > > > > -----Original Message----- > > From: Seven Du > > To: FreeSWITCH Users Help > > Sent: Sat, May 25, 2013 12:52 pm > > Subject: Re: [Freeswitch-users] dbh:query - insert id > > > > Maybe try the RETURNING clause ? > > > > -- > > Seven Du > > http://www.freeswitch.org.cn > > http://about.me/dujinfang > > http://www.dujinfang.com > > > > Sent with Sparrow > > > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: > >> Hello All > >> > >> How to get the id value after insert a record a record using dbh:query > >> > >> _table_a - columns_. > >> > >> id - auto increment > >> field1 > >> field2 > >> > >> > >> dbh:query("insert into table_a ( field1,field2) values ('11','Test')") > >> > >> > >> After insert how to get the table_a - id value for the inserted record? > >> > >> Thanks > >> Lloyd > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130525/155bfa0d/attachment.html From nneul at mst.edu Sat May 25 19:29:54 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Sat, 25 May 2013 10:29:54 -0500 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <51A0BA53.4030007@quentustech.com> <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> Message-ID: <51A0D8F2.4090808@mst.edu> Looking at the code, I'm not seeing anything that would do this. Affected rows is directly implemented as part of ODBC, that's why it's easily available. You could probably do a subsequent select on last_insert_id(), but there is a high risk of race condition - some other thread could use the dbh driver in the intervening time. I don't believe that the last insert id is returned automatically on any of the mysql api (underlying calls not FS) calls, and then you have the odbc layer on top of that. The way I've seen some other packages do it is to implement a mysql specific sql execute call that runs the statement, and then immediately runs a select last_insert_id() and returns it. -- Nathan On 05/25/2013 10:01 AM, Lloyd Aloysius wrote: > William, > > Thank you for the reply. Question was related to freeswitch dbh driver. Not related to sql specific. > > Most of the SQL Driver have a method to get the id after the insert record. My questions was does freeswitch dbh have > any method to get the id? > > For Example dbh have a method dbh:affected_rows() for insert , update. > > Like this do we have a method for ID? > > Thanks > Lloyd > * -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From bdfoster at davri.com Sat May 25 19:46:09 2013 From: bdfoster at davri.com (Brian Foster) Date: Sat, 25 May 2013 11:46:09 -0400 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: Debian FTW! - BDF On May 25, 2013 10:49 AM, "Ken Rice" wrote: > I actually dropped using Centos6 and moved to debian to get both timerfd > and to get performance at the same time. And things started working much > nicer... Not sure if they even fixed the performance issues on centos tho > > > On 5/25/13 4:52 AM, "Yuriy Nasida" wrote: > > Tamas, > > I think you can find info about performance problems with CentOS 6 on > jira. > For example. http://jira.freeswitch.org/browse/FS-4291 > > We also wanted to use timerfd (without virtualization) and made a moving > to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen > sometimes. As a result we had to move on centos 5.9 again. > > I would like to join issue. how can I be sure that timerfd is used? > > Regards, > Yuriy > > ------------------------------ > Date: Sat, 25 May 2013 07:52:28 +0200 > From: jalsot at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] OpenVZ tuning tips > > We've tried as root user but as I remember it was not able to set up the > priority but will recheck to be sure. Another interesting thing is that > when I tried to set a running process scheduler with chrt, got operation > not permitted (as root of course), so I guess, something has to be tuned in > the VE or on the host. We will try on bare metal centos6+ovz on the host. > We try the latest centos6 with latest openvz kernel (due to security > requirements we run on latest stable kernel and OS versions). > Could you give some info about those horrible performance problems to let > us check whether it still applies or not? (we've found only mysql create > table performance degradation due to ext4 - where the solution could be > barrier=0 yet, no other problems). > Actually, how can I be sure that timerfd is used? strace? I'm nearly sure > that timerfd works fine in FS. > > Yep, it would be much simpler without virtualization, and much harder from > another perspective. Probably lxc, kvm and xen aren't much better regarding > realtime stuff... > > Regards, > Tamas > > > On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > It will only work running as root I believe because it needs high privs to > do realtime. > If you do use centOS make sure its the latest rev of cent6, we have some > horrible performance problems on the earlier revs and I don't know if they > were resolved. > Just make sure your kernels are as new as you can get them on the real > host to avoid years of kernel performance bugs and that it has support for > timerfd otherwise your VE would compile with timerfd support but not have > actual access to the real syscalls for it in the host. > > That's why we try not to recommend virtual stuff in general as it takes > some very careful setups and its hard to support from our standpoint when > people run into issues. > > > > > > On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky > wrote: > > Hello, > > Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on > host and Ubuntu 10.04 LTS in VE. > Do you know maybe how to allow realtime priority in the VE for FS? Running > FS with -rp does not set the scheduler. > strace says, sched_setscheduler operation permitted, so SCHED_FF is not > set. Tried to run as root and/or use ulimit -r option, but cannot run FS > with tuned priorities.We guess, some thing missing in the host/VE > configuraton. > Any idea? > > Br, > Tamas > > > On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: > > Im using 2.6.32 on all my boxes ... > > One thing that has me thinking, are there any tweaks to get MSI-X working > best it can ? ( with proxmox ) > there seems to be a strong bias towards one CPU for all interrupts. > > I could be wrong, but its something I think ive seen, and didnt see any > clear suggestions on. > > Jay > > > > > > On 23 May 2013 01:12, Anthony Minessale > wrote: > > 2.6.25 or newer to get timerfd support. > > > > On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: > > On 5/22/13, Anthony Minessale wrote: > > You should consider centos6 or debian stable. Make sure the host kernel > is > > very new to get maximum results. > > Tony, do you mean "very new kernel" means 3.2.xx kernel? > > Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not > possible. And that is what CentOS6 offers, too. > > However, I installed FS as openvz guest, it works fine for outgoing, > but not DNAT works for incoming connections even after throroughly > following > http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. > > Just my two cents. > > > > > > > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky > wrote: > > > >> Hello, > >> > >> Do you have any recommendations regarding how to set up correctly (for > >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to > >> tuneup > >> the system to be rock solid? > >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with > FS > >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that > >> something is not well configured around timers, however mod_posix_timer > >> did > >> not help anything (running FS with -rp). We use our own bare metal and > >> can > >> reproduce those delatas eirher when only one VE is on the HW. > >> Maybe time to check out centos6 with openvz? > >> > >> Any idea, recommendation, experience can be very helpful. > >> > >> Regards, > >> Jalsot > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > >> > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > mailto:MSN%3Aanthony_minessale at hotmail.com> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com < > mailto:PAYPAL%3Aanthony.minessale at gmail.com> > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > mailto:sip%3A888 at conference.freeswitch.org> > > > googletalk:conf+888 at conference.freeswitch.org < > mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> > > > pstn:+19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130525/af28b69c/attachment-0001.html From j_mj at aol.com Sat May 25 19:52:36 2013 From: j_mj at aol.com (John M) Date: Sat, 25 May 2013 11:52:36 -0400 (EDT) Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <51A0D8F2.4090808@mst.edu> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <51A0BA53.4030007@quentustech.com> <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> <51A0D8F2.4090808@mst.edu> Message-ID: <8D02777031594A7-18C4-1A319@webmail-m236.sysops.aol.com> Hi Nathan, Yes, generally the current connection can make a select query and get the last inserted id and return that. It's possible, as suggested by Daniel earlier that doing a subsequent query for 'select last_insert_id()' may be the best way but that will be assuming the db connection doesn't refresh as that could potentially return the last inserted id of a concurrent channel. For me, I have implemented a solution where I have an extra column where I insert the channel uuid along with the data I insert as well. Then I make a select query to the db collecting the id where the uuid matches the current channel uuid and order by uuid desc. My expectation/hope is that I won't get 2 uuid's generated by freeswitch within a few seconds and should be able to rely on the id coming back as being the last inserted id for the insert query I had just made. Cheers, -Jm -----Original Message----- From: Nathan Neulinger To: FreeSWITCH Users Help Sent: Sun, May 26, 2013 1:35 am Subject: Re: [Freeswitch-users] dbh:query - insert id Looking at the code, I'm not seeing anything that would do this. Affected rows is directly implemented as part of ODBC, that's why it's easily available. You could probably do a subsequent select on last_insert_id(), but there is a high risk of race condition - some other thread could use the dbh driver in the intervening time. I don't believe that the last insert id is returned automatically on any of the mysql api (underlying calls not FS) calls, and then you have the odbc layer on top of that. The way I've seen some other packages do it is to implement a mysql specific sql execute call that runs the statement, and then immediately runs a select last_insert_id() and returns it. -- Nathan On 05/25/2013 10:01 AM, Lloyd Aloysius wrote: > William, > > Thank you for the reply. Question was related to freeswitch dbh driver. Not related to sql specific. > > Most of the SQL Driver have a method to get the id after the insert record. My questions was does freeswitch dbh have > any method to get the id? > > For Example dbh have a method dbh:affected_rows() for insert , update. > > Like this do we have a method for ID? > > Thanks > Lloyd > * -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130525/39e9f054/attachment.html From 8f27e956 at gmail.com Sat May 25 20:50:38 2013 From: 8f27e956 at gmail.com (S. Scott) Date: Sat, 25 May 2013 12:50:38 -0400 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D02777031594A7-18C4-1A319@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <51A0BA53.4030007@quentustech.com> <8D0276A8C7F27D1-18C4-19F17@webmail-m236.sysops.aol.com> <51A0D8F2.4090808@mst.edu> <8D02777031594A7-18C4-1A319@webmail-m236.sysops.aol.com> Message-ID: <9166475547196876026@unknownmsgid> Where the running DB engine supports a stored procedure, it is straightforward to implement the ID capture regardless of the connection abstraction (e.g ODBC). Unfortunately, sqlite3 is stored procedures challenged. Try reading, http://chriswolf.heroku.com/articles/2011/01/26/adding-stored-procedures-to-sqlite I have used his prescription successfully once. ????? iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors ? Thought of the Day ? "With all this manure, there must be a pony in here somewhere.? On 2013-05-25, at 11:56, John M wrote: Hi Nathan, Yes, generally the current connection can make a select query and get the last inserted id and return that. It's possible, as suggested by Daniel earlier that doing a subsequent query for 'select last_insert_id()' may be the best way but that will be assuming the db connection doesn't refresh as that could potentially return the last inserted id of a concurrent channel. For me, I have implemented a solution where I have an extra column where I insert the channel uuid along with the data I insert as well. Then I make a select query to the db collecting the id where the uuid matches the current channel uuid and order by uuid desc. My expectation/hope is that I won't get 2 uuid's generated by freeswitch within a few seconds and should be able to rely on the id coming back as being the last inserted id for the insert query I had just made. Cheers, -Jm -----Original Message----- From: Nathan Neulinger To: FreeSWITCH Users Help Sent: Sun, May 26, 2013 1:35 am Subject: Re: [Freeswitch-users] dbh:query - insert id Looking at the code, I'm not seeing anything that would do this. Affected rows is directly implemented as part of ODBC, that's why it's easily available. You could probably do a subsequent select on last_insert_id(), but there is a high risk of race condition - some other thread could use the dbh driver in the intervening time. I don't believe that the last insert id is returned automatically on any of the mysql api (underlying calls not FS) calls, and then you have the odbc layer on top of that. The way I've seen some other packages do it is to implement a mysql specific sql execute call that runs the statement, and then immediately runs a select last_insert_id() and returns it. -- Nathan On 05/25/2013 10:01 AM, Lloyd Aloysius wrote: > William, > > Thank you for the reply. Question was related to freeswitch dbh driver. Not related to sql specific. > > Most of the SQL Driver have a method to get the id after the insert record. My questions was does freeswitch dbh have > any method to get the id? > > For Example dbh have a method dbh:affected_rows() for insert , update. > > Like this do we have a method for ID? > > Thanks > Lloyd > * -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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/20130525/2e8aec71/attachment.html From fdelawarde at wirelessmundi.com Sat May 25 21:11:56 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Sat, 25 May 2013 19:11:56 +0200 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> Message-ID: <1369501916.15328.2.camel@salon.delawarde.com> Thanks to Steven and William for mentioning the RETURNING clause. I checked it out and learned something cool about PostgreSQL, so to me there's no problem with one liners like that. About MySQL's last_insert_id(), it is fine to use because it returns the last id for that particular connection/dbh handler, so you shouldn't need to worry about race as long as you use it with the same handler right after your insert. For other DB where race could be an issue (ex: sqlite3), you could always wrap it in a transaction to be sure: begin transaction; insert ; select last_insert_id(); end transaction; Fran?ois. On Sat, 2013-05-25 at 04:01 -0400, John M wrote: > Hi Daniel, > > Thanks for your description, it is much appreciated. :-) > > 5 word one liners from people too lazy to explain properly would > really be best if they didn't reply at all. > > Cheers, thanks again. > > -Jm > > > > > > > -----Original Message----- > From: Daniel Ivanov > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 5:57 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > It is true that the luasql driver is overly basic and poorly > documented . Unfortunately mysql doesn't support RETURNING clause like > pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); > right after the insert query. I cannot guarantee it works due to the > unknown nature(to me that is) of the luasql transaction handling, but > it should keep a transaction open for as long as a db handler lives. > > On May 25, 2013 7:03 AM, "John M" wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find > it documented anywhere. > > Instead of being cryptic, if you know the answer won't you > please help by explaining what the RETURNING clause is and how > to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch > community. > > -Jm > > > > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > Maybe try the RETURNING clause ? > > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > > Sent with Sparrow > > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: > > Hello All > > > > > > How to get the id value after insert a record a record using > > dbh:query > > > > > > table_a - columns. > > > > > > id - auto increment > > field1 > > field2 > > > > > > > > > > dbh:query("insert into table_a ( field1,field2) values > > ('11','Test')") > > > > > > > > > > After insert how to get the table_a - id value for the > > inserted record? > > > > > > Thanks > > Lloyd > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From william.king at quentustech.com Sat May 25 22:30:22 2013 From: william.king at quentustech.com (William King) Date: Sat, 25 May 2013 11:30:22 -0700 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <1369501916.15328.2.camel@salon.delawarde.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> <1369501916.15328.2.camel@salon.delawarde.com> Message-ID: <51A1033E.6010901@quentustech.com> Unfortunately you may hit a problem with last_insert_id() because FS uses connection pooling, and statement batches(at least for some select statements). Someone could check the lua wrapper code to see how the dbh handler code is implemented. 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 05/25/2013 10:11 AM, Fran?ois wrote: > Thanks to Steven and William for mentioning the RETURNING clause. I > checked it out and learned something cool about PostgreSQL, so to me > there's no problem with one liners like that. > > About MySQL's last_insert_id(), it is fine to use because it returns the > last id for that particular connection/dbh handler, so you shouldn't > need to worry about race as long as you use it with the same handler > right after your insert. > > For other DB where race could be an issue (ex: sqlite3), you could > always wrap it in a transaction to be sure: > > begin transaction; > insert ; > select last_insert_id(); > end transaction; > > Fran?ois. > > On Sat, 2013-05-25 at 04:01 -0400, John M wrote: >> Hi Daniel, >> >> Thanks for your description, it is much appreciated. :-) >> >> 5 word one liners from people too lazy to explain properly would >> really be best if they didn't reply at all. >> >> Cheers, thanks again. >> >> -Jm >> >> >> >> >> >> >> -----Original Message----- >> From: Daniel Ivanov >> To: FreeSWITCH Users Help >> Sent: Sat, May 25, 2013 5:57 pm >> Subject: Re: [Freeswitch-users] dbh:query - insert id >> >> It is true that the luasql driver is overly basic and poorly >> documented . Unfortunately mysql doesn't support RETURNING clause like >> pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); >> right after the insert query. I cannot guarantee it works due to the >> unknown nature(to me that is) of the luasql transaction handling, but >> it should keep a transaction open for as long as a db handler lives. >> >> On May 25, 2013 7:03 AM, "John M" wrote: >> Hi Seven Du, >> >> I'd really like to know if this is possible too, couldn't find >> it documented anywhere. >> >> Instead of being cryptic, if you know the answer won't you >> please help by explaining what the RETURNING clause is and how >> to use it? >> >> Does it somehow return mysql_insert_id()? >> >> How should we use it. >> >> You help is invaluable and is contributing to the freeswitch >> community. >> >> -Jm >> >> >> >> >> >> -----Original Message----- >> From: Seven Du >> To: FreeSWITCH Users Help >> >> Sent: Sat, May 25, 2013 12:52 pm >> Subject: Re: [Freeswitch-users] dbh:query - insert id >> >> Maybe try the RETURNING clause ? >> >> >> -- >> Seven Du >> http://www.freeswitch.org.cn >> http://about.me/dujinfang >> http://www.dujinfang.com >> >> >> Sent with Sparrow >> >> >> On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: >> > Hello All >> > >> > >> > How to get the id value after insert a record a record using >> > dbh:query >> > >> > >> > table_a - columns. >> > >> > >> > id - auto increment >> > field1 >> > field2 >> > >> > >> > >> > >> > dbh:query("insert into table_a ( field1,field2) values >> > ('11','Test')") >> > >> > >> > >> > >> > After insert how to get the table_a - id value for the >> > inserted record? >> > >> > >> > Thanks >> > Lloyd >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From krice at freeswitch.org Sat May 25 22:51:12 2013 From: krice at freeswitch.org (Ken Rice) Date: Sat, 25 May 2013 13:51:12 -0500 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Message-ID: I don?t know if mysql even supports the returning keywork... The way RETURNING works is its an addon to your insert statements INSERT INTO table_foo (col_a, col_b) values (?foo?, ?bar?) RETURNING row_id; Where row_id is an auto incrementing field such as a SERIAL (sequence) in PGSQL or an auto-increment field in mySQL. While this is an ?insert? per say, it looks more like a ?select? when executing as data is returned On 5/24/13 10:59 PM, "John M" wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it documented > anywhere. > > Instead of being cryptic, if you know the answer won't you please help by > explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > > Maybe try the RETURNING clause ? > -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130525/9c3f2178/attachment.html From sertys at gmail.com Sun May 26 09:01:55 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 26 May 2013 07:01:55 +0200 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: References: <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> Message-ID: It doesn't support it. And still we're looking at luasql driver, not freeswitch or odbc implementations. Just use the LAST_INSERT_ID() function. If it returns weird things, disable autocommit and build your own transactions to be certain you get the right value. And as addition transaction != session != connection when it comes to modern database driving. On May 25, 2013 9:55 PM, "Ken Rice" wrote: > I don?t know if mysql even supports the returning keywork... > > The way RETURNING works is its an addon to your insert statements > > INSERT INTO table_foo (col_a, col_b) values (?foo?, ?bar?) RETURNING > row_id; > > Where row_id is an auto incrementing field such as a SERIAL (sequence) in > PGSQL or an auto-increment field in mySQL. > > While this is an ?insert? per say, it looks more like a ?select? when > executing as data is returned > > > > On 5/24/13 10:59 PM, "John M" wrote: > > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it > documented anywhere. > > Instead of being cryptic, if you know the answer won't you please help by > explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > > Maybe try the RETURNING clause ? > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/1485b098/attachment.html From sertys at gmail.com Sun May 26 10:19:55 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 26 May 2013 08:19:55 +0200 Subject: [Freeswitch-users] SRTP media handling survey behaviour. Message-ID: I wanted to implement a mixed mode of srtp and zrtp crypto capabilities across our network and i was facing several issues. But succeeded nonetheless. What kicked me in the balls hardest was rfc3713(taken from memory) compliance of the UAs. Whenever i wanted to set SRTP to optional on the devices sent both avp and savp lines along with the according keys in the a=crypto lines in the sdp and the zrtp-hash too. If for a reason i wanted to force ZRTP on the channel, i was supposed to set sip_secure_media to false and that would disable srtp on the b-leg and zrtp handshake would take place in p2p mode. I was using csipsimple(pjsip) though and fs was just stripping the savp from the sdp and still passing the a=crypto lines. It turned out greedy and still started talking srtp instead of zrtp. I patched it myself to strip the a=crypto too if sip_secure_media is false and it worked like a german. I wanted to know if anyone has observed how other UAs handle the srtp negotiation and if it's worth proposing my patch as an improvement on the JIRA. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130526/ca5e551e/attachment-0001.html From chris at the-brannons.com Fri May 24 19:14:51 2013 From: chris at the-brannons.com (Chris Brannon) Date: Fri, 24 May 2013 08:14:51 -0700 Subject: [Freeswitch-users] phone hangs when placing internal calls Message-ID: <87vc68z9xw.fsf@mushroom.localdomain> Hello, I'm very new to freeswitch and telephony in general. I've managed to register phones with my server, but when I place test calls, the phone hangs, even though fs_cli tells me that the call was completed successfully. Also, everything looks fine when I run sip trace. FWIW, I've tried both linphone on Linux and csipsimple on Android. All of this is taking place on my LAN, so NAT isn't an issue. I started looking at traffic with tcpdump, and here's what I found. The phone authenticates to FreeSwitch and initiates the call. FreeSwitch sends a "100 trying" SIP message, followed by a UDP packet containing what appears to be SDP. It has no SIP header. This is followed by RTP from FreeSwitch to the phone. There is never a "200 OK" SIP message. Next, I ran linphone in debugging mode. I'll include a link to the log, in the hope that it will be helpful. Apparently, linphone isn't parsing the packet after the 100 trying message. Any idea what is going on here? http://the-brannons.com/linphone.log -- Chris From eoaddai at gmail.com Sat May 25 12:01:41 2013 From: eoaddai at gmail.com (eoaddai) Date: Sat, 25 May 2013 01:01:41 -0700 (PDT) Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION Message-ID: <1369468901841-7591099.post@n2.nabble.com> Hi, my first time of posting stuff here. I really need help. I keep on not getting my calls go through freeswitch to my goip gateway. The following is the freeswitch log with siptrace turned on i get. Please help me: 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [origination_caller_id_number]=[0264370536] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_from]=[0264370536] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_to]=[0267577771] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [plivo_app]=[true] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [originate_timeout]=[60] 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 (sofia/external/0267577771) State Change CS_NEW -> CS_INIT 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0267577771) Running State Change CS_INIT 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0267577771) State INIT 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 sofia/external/0267577771 SOFIA INIT 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 s=FreeSWITCH c=IN IP4 10.10.50.1 t=0 0 m=audio 30174 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0267577771) State INIT going to sleep send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: ------------------------------------------------------------------------ INVITE sip:0267577771 at 10.10.50.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N Max-Forwards: 70 From: "" ;tag=DXjN0tK6mcc5S To: Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 201 X-FS-Support: update_display,send_info Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 s=FreeSWITCH c=IN IP4 10.10.50.1 t=0 0 m=audio 30174 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0267577771) Running State Change CS_ROUTING 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 (sofia/external/0267577771) State ROUTING 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 sofia/external/0267577771 SOFIA ROUTING 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 (sofia/external/0267577771) State ROUTING going to sleep 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 (sofia/external/0267577771) State CONSUME_MEDIA 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 (sofia/external/0267577771) State CONSUME_MEDIA going to sleep 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel sofia/external/0267577771 entering state [calling][0] recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: ------------------------------------------------------------------------ SIP/2.0 606 Not Acceptable Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N From: "" ;tag=DXjN0tK6mcc5S To: ;tag=1662509363 Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 INVITE User-Agent: dble Content-Length: 0 ------------------------------------------------------------------------ send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: ------------------------------------------------------------------------ ACK sip:0267577771 at 10.10.50.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N Max-Forwards: 70 From: "" ;tag=DXjN0tK6mcc5S To: ;tag=1662509363 Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 ACK Content-Length: 0 ------------------------------------------------------------------------ 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel sofia/external/0267577771 entering state [terminated][606] 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal sofia/external/0267577771 [KILL] 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0267577771) Running State Change CS_HANGUP 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 (sofia/external/0267577771) State HANGUP 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 sofia/external/0267577771 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 (sofia/external/0267577771) State HANGUP going to sleep 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 (sofia/external/0267577771) Callstate Change DOWN -> HANGUP 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0267577771) Running State Change CS_REPORTING 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 (sofia/external/0267577771) State REPORTING 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 sofia/external/0267577771 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 (sofia/external/0267577771) State REPORTING going to sleep 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0267577771 [BREAK] 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 (sofia/external/0267577771) Locked, Waiting on external entities 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session 3 (sofia/external/0267577771) Ended 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/0267577771 [CS_DESTROY] 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 (sofia/external/0267577771) Callstate Change HANGUP -> DOWN 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 (sofia/external/0267577771) Running State Change CS_DESTROY 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 (sofia/external/0267577771) State DESTROY 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 sofia/external/0267577771 SOFIA DESTROY 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 sofia/external/0267577771 Standard DESTROY 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 (sofia/external/0267577771) State DESTROY going to sleep 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing global variables 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [origination_caller_id_number]=[0264370536] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_from]=[0264370536] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_to]=[0249230704] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [plivo_app]=[true] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [originate_timeout]=[60] 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable [ignore_early_media]=[true] 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 (sofia/external/0249230704) State Change CS_NEW -> CS_INIT 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0249230704) Running State Change CS_INIT 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0249230704) State INIT 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 sofia/external/0249230704 SOFIA INIT 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: v=0 o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 s=FreeSWITCH c=IN IP4 10.10.50.1 t=0 0 m=audio 19250 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 (sofia/external/0249230704) State INIT going to sleep send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: ------------------------------------------------------------------------ INVITE sip:0249230704 at 10.10.50.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH Max-Forwards: 70 From: "" ;tag=e6Be2N49HN2QN To: Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 201 X-FS-Support: update_display,send_info Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 s=FreeSWITCH c=IN IP4 10.10.50.1 t=0 0 m=audio 19250 RTP/AVP 0 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0249230704) Running State Change CS_ROUTING 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 (sofia/external/0249230704) State ROUTING 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 sofia/external/0249230704 SOFIA ROUTING 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 (sofia/external/0249230704) State ROUTING going to sleep 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 (sofia/external/0249230704) State CONSUME_MEDIA 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 (sofia/external/0249230704) State CONSUME_MEDIA going to sleep 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel sofia/external/0249230704 entering state [calling][0] recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: ------------------------------------------------------------------------ SIP/2.0 606 Not Acceptable Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH From: "" ;tag=e6Be2N49HN2QN To: ;tag=633680086 Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 INVITE User-Agent: dble Content-Length: 0 ------------------------------------------------------------------------ send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: ------------------------------------------------------------------------ ACK sip:0249230704 at 10.10.50.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH Max-Forwards: 70 From: "" ;tag=e6Be2N49HN2QN To: ;tag=633680086 Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 CSeq: 44398096 ACK Content-Length: 0 ------------------------------------------------------------------------ 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel sofia/external/0249230704 entering state [terminated][606] 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal sofia/external/0249230704 [KILL] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0249230704) Running State Change CS_HANGUP 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 (sofia/external/0249230704) State HANGUP 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 sofia/external/0249230704 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 (sofia/external/0249230704) State HANGUP going to sleep 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 (sofia/external/0249230704) Callstate Change DOWN -> HANGUP 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 (sofia/external/0249230704) Running State Change CS_REPORTING 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 (sofia/external/0249230704) State REPORTING 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 sofia/external/0249230704 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 (sofia/external/0249230704) State REPORTING going to sleep 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/0249230704 [BREAK] 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 (sofia/external/0249230704) Locked, Waiting on external entities 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session 4 (sofia/external/0249230704) Ended 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/0249230704 [CS_DESTROY] 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 (sofia/external/0249230704) Callstate Change HANGUP -> DOWN 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 (sofia/external/0249230704) Running State Change CS_DESTROY 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 (sofia/external/0249230704) State DESTROY 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 sofia/external/0249230704 SOFIA DESTROY 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 sofia/external/0249230704 Standard DESTROY 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 (sofia/external/0249230704) State DESTROY going to sleep -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ehermouet at bluetel.fr Sun May 26 12:37:59 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Sun, 26 May 2013 10:37:59 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> Message-ID: <8fee6197-5f94-418c-a298-f0004c469d95@email.android.com> Nobody know the good change to passe rfc2833 to info ? Tks advance Michael Jerris a ?crit?: >if you see the rtp events going out but you don't see it having any >affect, try asking your provider? > >Mike > >On May 24, 2013, at 5:31 AM, ehermouet at bluetel.fr wrote: > >> After some hours i foudn rtp event with wireshark. >> >> RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) >> but no result on ivr outbound... >> >> >> >> >> Le 2013-05-23 18:06, Michael Jerris a ?crit : >>> Have you looked at it to see if it is sending the dtmf? >>> >>> On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: >>> >>>> you can found here my tcpdump file >>>> >>>> http://bluetelconnect.fr/tcpdump.log >>>> >>>> tks advance Michael >>>> >>>> >>>> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>>>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>>>> review of what you have posted so far. >>>>> >>>>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>>>> >>>>>> pcap. >>>>>> >>>>>> i send you the xml file and log in my previous email... because i >>>>>> see >>>>>> problem sometime... i'm sure i have error on my xml file. can you >>>>>> check >>>>>> it. ? >>>>>> >>>>>> tks >>>>>> >>>>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>>>> How do you use what? >>>>>>> >>>>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>>>> >>>>>>>> how do you use it without interface ? it's server with only ssh >>>>>>>> access. >>>>>>>> tks >>>>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>>>> this log does not seem to have a complete call let alone any >>>>>>>>> attempt >>>>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>>>> said, >>>>>>>>> its >>>>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>>>> flowing >>>>>>>>> ? >>>>>>>>> >>>>>>>>> Mike >>>>>>>>> >>>>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>>>> >>>>>>>>>> Yes >>>>>>>>>> >>>>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>>>> >>>>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>>>> negotiation >>>>>>>>>>> settings? >>> >>> >>> >_________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >> >> >_________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 Hermouet Erwan Responsable technique Bluetel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130526/b95c893c/attachment-0001.html From sertys at gmail.com Sun May 26 14:25:19 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Sun, 26 May 2013 12:25:19 +0200 Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION In-Reply-To: <1369468901841-7591099.post@n2.nabble.com> References: <1369468901841-7591099.post@n2.nabble.com> Message-ID: Maybe you're not sending them the right codecs or trying to run a feature they don't have. Revisit your vars.xml and sip_profiles .xml. . Ultimately contact the provider to ask them what youre doing wrong in your sdps. On May 26, 2013 11:26 AM, "eoaddai" wrote: > Hi, my first time of posting stuff here. I really need help. > I keep on not getting my calls go through freeswitch to my goip gateway. > The > following is the freeswitch log with siptrace turned on i get. Please help > me: > > 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing > global variables > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [origination_caller_id_number]=[0264370536] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_from]=[0264370536] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_to]=[0267577771] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_app]=[true] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [originate_timeout]=[60] > 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable > [ignore_early_media]=[true] > 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel > sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] > 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 > (sofia/external/0267577771) State Change CS_NEW -> CS_INIT > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0267577771) Running State Change CS_INIT > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/0267577771) State INIT > 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 sofia/external/0267577771 > SOFIA INIT > 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: > v=0 > o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 > s=FreeSWITCH > c=IN IP4 10.10.50.1 > t=0 0 > m=audio 30174 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 > (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/0267577771) State INIT going to sleep > send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: > ------------------------------------------------------------------------ > INVITE sip:0267577771 at 10.10.50.10 SIP/2.0 > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N > Max-Forwards: 70 > From: "" ;tag=DXjN0tK6mcc5S > To: > Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 201 > X-FS-Support: update_display,send_info > Remote-Party-ID: > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 > s=FreeSWITCH > c=IN IP4 10.10.50.1 > t=0 0 > m=audio 30174 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0267577771) Running State Change CS_ROUTING > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/0267577771) State ROUTING > 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 > sofia/external/0267577771 > SOFIA ROUTING > 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 > (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/0267577771) State ROUTING going to sleep > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/0267577771) State CONSUME_MEDIA > 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/0267577771) State CONSUME_MEDIA going to sleep > 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel > sofia/external/0267577771 entering state [calling][0] > recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: > ------------------------------------------------------------------------ > SIP/2.0 606 Not Acceptable > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N > From: "" ;tag=DXjN0tK6mcc5S > To: ;tag=1662509363 > Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 INVITE > User-Agent: dble > Content-Length: 0 > > ------------------------------------------------------------------------ > send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: > ------------------------------------------------------------------------ > ACK sip:0267577771 at 10.10.50.10 SIP/2.0 > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N > Max-Forwards: 70 > From: "" ;tag=DXjN0tK6mcc5S > To: ;tag=1662509363 > Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel > sofia/external/0267577771 entering state [terminated][606] > 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup > sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 Originate > Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal > sofia/external/0267577771 [KILL] > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0267577771) Running State Change CS_HANGUP > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/0267577771) State HANGUP > 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel > sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 > sofia/external/0267577771 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/0267577771) State HANGUP going to sleep > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 > (sofia/external/0267577771) Callstate Change DOWN -> HANGUP > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0267577771) Running State Change CS_REPORTING > 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/0267577771) State REPORTING > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 > sofia/external/0267577771 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/0267577771) State REPORTING going to sleep > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0267577771 [BREAK] > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 > (sofia/external/0267577771) Locked, Waiting on external entities > 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session 3 > (sofia/external/0267577771) Ended > 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close > Channel > sofia/external/0267577771 [CS_DESTROY] > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 > (sofia/external/0267577771) Callstate Change HANGUP -> DOWN > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 > (sofia/external/0267577771) Running State Change CS_DESTROY > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 > (sofia/external/0267577771) State DESTROY > 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 > sofia/external/0267577771 > SOFIA DESTROY > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 > sofia/external/0267577771 Standard DESTROY > 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 > (sofia/external/0267577771) State DESTROY going to sleep > 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing > global variables > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [origination_caller_id_number]=[0264370536] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_from]=[0264370536] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_to]=[0249230704] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [plivo_app]=[true] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [originate_timeout]=[60] > 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable > [ignore_early_media]=[true] > 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel > sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] > 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 > (sofia/external/0249230704) State Change CS_NEW -> CS_INIT > 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0249230704) Running State Change CS_INIT > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/0249230704) State INIT > 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 sofia/external/0249230704 > SOFIA INIT > 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: > v=0 > o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 > s=FreeSWITCH > c=IN IP4 10.10.50.1 > t=0 0 > m=audio 19250 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 > (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 > (sofia/external/0249230704) State INIT going to sleep > send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: > ------------------------------------------------------------------------ > INVITE sip:0249230704 at 10.10.50.10 SIP/2.0 > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH > Max-Forwards: 70 > From: "" ;tag=e6Be2N49HN2QN > To: > Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 201 > X-FS-Support: update_display,send_info > Remote-Party-ID: > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 > s=FreeSWITCH > c=IN IP4 10.10.50.1 > t=0 0 > m=audio 19250 RTP/AVP 0 8 3 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > ------------------------------------------------------------------------ > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0249230704) Running State Change CS_ROUTING > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/0249230704) State ROUTING > 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 > sofia/external/0249230704 > SOFIA ROUTING > 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 > (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/0249230704) State ROUTING going to sleep > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/0249230704) State CONSUME_MEDIA > 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 > (sofia/external/0249230704) State CONSUME_MEDIA going to sleep > 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel > sofia/external/0249230704 entering state [calling][0] > recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: > ------------------------------------------------------------------------ > SIP/2.0 606 Not Acceptable > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH > From: "" ;tag=e6Be2N49HN2QN > To: ;tag=633680086 > Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 INVITE > User-Agent: dble > Content-Length: 0 > > ------------------------------------------------------------------------ > send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: > ------------------------------------------------------------------------ > ACK sip:0249230704 at 10.10.50.10 SIP/2.0 > Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH > Max-Forwards: 70 > From: "" ;tag=e6Be2N49HN2QN > To: ;tag=633680086 > Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 > CSeq: 44398096 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel > sofia/external/0249230704 entering state [terminated][606] > 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup > sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] > 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal > sofia/external/0249230704 [KILL] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0249230704) Running State Change CS_HANGUP > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/0249230704) State HANGUP > 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel > sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 > sofia/external/0249230704 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/0249230704) State HANGUP going to sleep > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 > (sofia/external/0249230704) Callstate Change DOWN -> HANGUP > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/0249230704) Running State Change CS_REPORTING > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/0249230704) State REPORTING > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 > sofia/external/0249230704 Standard REPORTING, cause: > INCOMPATIBLE_DESTINATION > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/0249230704) State REPORTING going to sleep > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal > sofia/external/0249230704 [BREAK] > 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 > (sofia/external/0249230704) Locked, Waiting on external entities > 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 Originate > Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] > 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session 4 > (sofia/external/0249230704) Ended > 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close > Channel > sofia/external/0249230704 [CS_DESTROY] > 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 > (sofia/external/0249230704) Callstate Change HANGUP -> DOWN > 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 > (sofia/external/0249230704) Running State Change CS_DESTROY > 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 > (sofia/external/0249230704) State DESTROY > 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 > sofia/external/0249230704 > SOFIA DESTROY > 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 > sofia/external/0249230704 Standard DESTROY > 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 > (sofia/external/0249230704) State DESTROY going to sleep > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.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://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/20130526/461c6536/attachment-0001.html From steveayre at gmail.com Sun May 26 14:46:39 2013 From: steveayre at gmail.com (Steven Ayre) Date: Sun, 26 May 2013 11:46:39 +0100 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <51A1033E.6010901@quentustech.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> <1369501916.15328.2.camel@salon.delawarde.com> <51A1033E.6010901@quentustech.com> Message-ID: If you switch from luasql to freeswitch.Dbh then the freeswitch.Dbh object will give a single connection until you either call the release method or the variable goes out of scope. That means queries will happen on the same transaction. -Steve On 25 May 2013 19:30, William King wrote: > Unfortunately you may hit a problem with last_insert_id() because FS > uses connection pooling, and statement batches(at least for some select > statements). Someone could check the lua wrapper code to see how the dbh > handler code is implemented. > > 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 05/25/2013 10:11 AM, Fran?ois wrote: > > Thanks to Steven and William for mentioning the RETURNING clause. I > > checked it out and learned something cool about PostgreSQL, so to me > > there's no problem with one liners like that. > > > > About MySQL's last_insert_id(), it is fine to use because it returns the > > last id for that particular connection/dbh handler, so you shouldn't > > need to worry about race as long as you use it with the same handler > > right after your insert. > > > > For other DB where race could be an issue (ex: sqlite3), you could > > always wrap it in a transaction to be sure: > > > > begin transaction; > > insert ; > > select last_insert_id(); > > end transaction; > > > > Fran?ois. > > > > On Sat, 2013-05-25 at 04:01 -0400, John M wrote: > >> Hi Daniel, > >> > >> Thanks for your description, it is much appreciated. :-) > >> > >> 5 word one liners from people too lazy to explain properly would > >> really be best if they didn't reply at all. > >> > >> Cheers, thanks again. > >> > >> -Jm > >> > >> > >> > >> > >> > >> > >> -----Original Message----- > >> From: Daniel Ivanov > >> To: FreeSWITCH Users Help > >> Sent: Sat, May 25, 2013 5:57 pm > >> Subject: Re: [Freeswitch-users] dbh:query - insert id > >> > >> It is true that the luasql driver is overly basic and poorly > >> documented . Unfortunately mysql doesn't support RETURNING clause like > >> pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); > >> right after the insert query. I cannot guarantee it works due to the > >> unknown nature(to me that is) of the luasql transaction handling, but > >> it should keep a transaction open for as long as a db handler lives. > >> > >> On May 25, 2013 7:03 AM, "John M" wrote: > >> Hi Seven Du, > >> > >> I'd really like to know if this is possible too, couldn't find > >> it documented anywhere. > >> > >> Instead of being cryptic, if you know the answer won't you > >> please help by explaining what the RETURNING clause is and how > >> to use it? > >> > >> Does it somehow return mysql_insert_id()? > >> > >> How should we use it. > >> > >> You help is invaluable and is contributing to the freeswitch > >> community. > >> > >> -Jm > >> > >> > >> > >> > >> > >> -----Original Message----- > >> From: Seven Du > >> To: FreeSWITCH Users Help > >> > >> Sent: Sat, May 25, 2013 12:52 pm > >> Subject: Re: [Freeswitch-users] dbh:query - insert id > >> > >> Maybe try the RETURNING clause ? > >> > >> > >> -- > >> Seven Du > >> http://www.freeswitch.org.cn > >> http://about.me/dujinfang > >> http://www.dujinfang.com > >> > >> > >> Sent with Sparrow > >> > >> > >> On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: > >> > Hello All > >> > > >> > > >> > How to get the id value after insert a record a record using > >> > dbh:query > >> > > >> > > >> > table_a - columns. > >> > > >> > > >> > id - auto increment > >> > field1 > >> > field2 > >> > > >> > > >> > > >> > > >> > dbh:query("insert into table_a ( field1,field2) values > >> > ('11','Test')") > >> > > >> > > >> > > >> > > >> > After insert how to get the table_a - id value for the > >> > inserted record? > >> > > >> > > >> > Thanks > >> > Lloyd > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.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 > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/f70ae38c/attachment-0001.html From jalsot at gmail.com Sun May 26 21:09:45 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Sun, 26 May 2013 19:09:45 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: Hello Ken, Do you use any kind of virtualizaztion/separation on Debian? We would like to keep lighweight 'virtualization' with openvz or maybe lxc (or any other idea?). Any experience with that on Debian? Is Debian better suited for FS than Ubuntu LTS (e.g. 12.04)? It seems, we have to find out where to go from latest centos5+ovz+ubuntu 10.04 in VE... T. On Sat, May 25, 2013 at 4:42 PM, Ken Rice wrote: > I actually dropped using Centos6 and moved to debian to get both timerfd > and to get performance at the same time. And things started working much > nicer... Not sure if they even fixed the performance issues on centos tho > > > > On 5/25/13 4:52 AM, "Yuriy Nasida" wrote: > > Tamas, > > I think you can find info about performance problems with CentOS 6 on > jira. > For example. http://jira.freeswitch.org/browse/FS-4291 > > We also wanted to use timerfd (without virtualization) and made a moving > to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen > sometimes. As a result we had to move on centos 5.9 again. > > I would like to join issue. how can I be sure that timerfd is used? > > Regards, > Yuriy > > ------------------------------ > Date: Sat, 25 May 2013 07:52:28 +0200 > From: jalsot at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] OpenVZ tuning tips > > We've tried as root user but as I remember it was not able to set up the > priority but will recheck to be sure. Another interesting thing is that > when I tried to set a running process scheduler with chrt, got operation > not permitted (as root of course), so I guess, something has to be tuned in > the VE or on the host. We will try on bare metal centos6+ovz on the host. > We try the latest centos6 with latest openvz kernel (due to security > requirements we run on latest stable kernel and OS versions). > Could you give some info about those horrible performance problems to let > us check whether it still applies or not? (we've found only mysql create > table performance degradation due to ext4 - where the solution could be > barrier=0 yet, no other problems). > Actually, how can I be sure that timerfd is used? strace? I'm nearly sure > that timerfd works fine in FS. > > Yep, it would be much simpler without virtualization, and much harder from > another perspective. Probably lxc, kvm and xen aren't much better regarding > realtime stuff... > > Regards, > Tamas > > > On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > It will only work running as root I believe because it needs high privs to > do realtime. > If you do use centOS make sure its the latest rev of cent6, we have some > horrible performance problems on the earlier revs and I don't know if they > were resolved. > Just make sure your kernels are as new as you can get them on the real > host to avoid years of kernel performance bugs and that it has support for > timerfd otherwise your VE would compile with timerfd support but not have > actual access to the real syscalls for it in the host. > > That's why we try not to recommend virtual stuff in general as it takes > some very careful setups and its hard to support from our standpoint when > people run into issues. > > > > > > On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky > wrote: > > Hello, > > Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on > host and Ubuntu 10.04 LTS in VE. > Do you know maybe how to allow realtime priority in the VE for FS? Running > FS with -rp does not set the scheduler. > strace says, sched_setscheduler operation permitted, so SCHED_FF is not > set. Tried to run as root and/or use ulimit -r option, but cannot run FS > with tuned priorities.We guess, some thing missing in the host/VE > configuraton. > Any idea? > > Br, > Tamas > > > On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: > > Im using 2.6.32 on all my boxes ... > > One thing that has me thinking, are there any tweaks to get MSI-X working > best it can ? ( with proxmox ) > there seems to be a strong bias towards one CPU for all interrupts. > > I could be wrong, but its something I think ive seen, and didnt see any > clear suggestions on. > > Jay > > > > > > On 23 May 2013 01:12, Anthony Minessale > wrote: > > 2.6.25 or newer to get timerfd support. > > > > On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: > > On 5/22/13, Anthony Minessale wrote: > > You should consider centos6 or debian stable. Make sure the host kernel > is > > very new to get maximum results. > > Tony, do you mean "very new kernel" means 3.2.xx kernel? > > Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not > possible. And that is what CentOS6 offers, too. > > However, I installed FS as openvz guest, it works fine for outgoing, > but not DNAT works for incoming connections even after throroughly > following > http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. > > Just my two cents. > > > > > > > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky > wrote: > > > >> Hello, > >> > >> Do you have any recommendations regarding how to set up correctly (for > >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to > >> tuneup > >> the system to be rock solid? > >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with > FS > >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that > >> something is not well configured around timers, however mod_posix_timer > >> did > >> not help anything (running FS with -rp). We use our own bare metal and > >> can > >> reproduce those delatas eirher when only one VE is on the HW. > >> Maybe time to check out centos6 with openvz? > >> > >> Any idea, recommendation, experience can be very helpful. > >> > >> Regards, > >> Jalsot > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > >> > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com < > mailto:MSN%3Aanthony_minessale at hotmail.com> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com < > mailto:PAYPAL%3Aanthony.minessale at gmail.com> > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org < > mailto:sip%3A888 at conference.freeswitch.org> > > > googletalk:conf+888 at conference.freeswitch.org < > mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> > > > pstn:+19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/08915a69/attachment.html From fdelawarde at wirelessmundi.com Sun May 26 21:18:28 2013 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=) Date: Sun, 26 May 2013 19:18:28 +0200 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> <1369501916.15328.2.camel@salon.delawarde.com> <51A1033E.6010901@quentustech.com> Message-ID: <1369588708.15328.6.camel@salon.delawarde.com> Indeed, sorry I forgot to mention that. Still in sqlite3 case, you need to wrap it in a transaction in case another thread get a lock on the db between your two queries. Fran?ois. On Sun, 2013-05-26 at 11:46 +0100, Steven Ayre wrote: > If you switch from luasql to freeswitch.Dbh then the freeswitch.Dbh > object will give a single connection until you either call the release > method or the variable goes out of scope. That means queries will > happen on the same transaction. > > > -Steve > > > On 25 May 2013 19:30, William King > wrote: > Unfortunately you may hit a problem with last_insert_id() > because FS > uses connection pooling, and statement batches(at least for > some select > statements). Someone could check the lua wrapper code to see > how the dbh > handler code is implemented. > > 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 05/25/2013 10:11 AM, Fran?ois wrote: > > Thanks to Steven and William for mentioning the RETURNING > clause. I > > checked it out and learned something cool about PostgreSQL, > so to me > > there's no problem with one liners like that. > > > > About MySQL's last_insert_id(), it is fine to use because it > returns the > > last id for that particular connection/dbh handler, so you > shouldn't > > need to worry about race as long as you use it with the same > handler > > right after your insert. > > > > For other DB where race could be an issue (ex: sqlite3), you > could > > always wrap it in a transaction to be sure: > > > > begin transaction; > > insert ; > > select last_insert_id(); > > end transaction; > > > > Fran?ois. > > > > On Sat, 2013-05-25 at 04:01 -0400, John M wrote: > >> Hi Daniel, > >> > >> Thanks for your description, it is much appreciated. :-) > >> > >> 5 word one liners from people too lazy to explain properly > would > >> really be best if they didn't reply at all. > >> > >> Cheers, thanks again. > >> > >> -Jm > >> > >> > >> > >> > >> > >> > >> -----Original Message----- > >> From: Daniel Ivanov > >> To: FreeSWITCH Users Help > > >> Sent: Sat, May 25, 2013 5:57 pm > >> Subject: Re: [Freeswitch-users] dbh:query - insert id > >> > >> It is true that the luasql driver is overly basic and > poorly > >> documented . Unfortunately mysql doesn't support RETURNING > clause like > >> pgsql and oracle. You should however try SELECT > LAST_INSERT_ID(); > >> right after the insert query. I cannot guarantee it works > due to the > >> unknown nature(to me that is) of the luasql transaction > handling, but > >> it should keep a transaction open for as long as a db > handler lives. > >> > >> On May 25, 2013 7:03 AM, "John M" wrote: > >> Hi Seven Du, > >> > >> I'd really like to know if this is possible too, > couldn't find > >> it documented anywhere. > >> > >> Instead of being cryptic, if you know the answer > won't you > >> please help by explaining what the RETURNING clause > is and how > >> to use it? > >> > >> Does it somehow return mysql_insert_id()? > >> > >> How should we use it. > >> > >> You help is invaluable and is contributing to the > freeswitch > >> community. > >> > >> -Jm > >> > >> > >> > >> > >> > >> -----Original Message----- > >> From: Seven Du > >> To: FreeSWITCH Users Help > >> > >> Sent: Sat, May 25, 2013 12:52 pm > >> Subject: Re: [Freeswitch-users] dbh:query - insert > id > >> > >> Maybe try the RETURNING clause ? > >> > >> > >> -- > >> Seven Du > >> http://www.freeswitch.org.cn > >> http://about.me/dujinfang > >> http://www.dujinfang.com > >> > >> > >> Sent with Sparrow > >> > >> > >> On Saturday, May 25, 2013 at 8:14 AM, Lloyd > Aloysius wrote: > >> > Hello All > >> > > >> > > >> > How to get the id value after insert a record a > record using > >> > dbh:query > >> > > >> > > >> > table_a - columns. > >> > > >> > > >> > id - auto increment > >> > field1 > >> > field2 > >> > > >> > > >> > > >> > > >> > dbh:query("insert into table_a ( field1,field2) > values > >> > ('11','Test')") > >> > > >> > > >> > > >> > > >> > After insert how to get the table_a - id value > for the > >> > inserted record? > >> > > >> > > >> > Thanks > >> > Lloyd > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > >> > > >> > > >> > > >> > 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 > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > >> > >> > >> 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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > >> > >> > >> 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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.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 > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From krice at freeswitch.org Sun May 26 22:12:32 2013 From: krice at freeswitch.org (Ken Rice) Date: Sun, 26 May 2013 13:12:32 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: References: Message-ID: <9BCB536D-E68D-4AD7-80CD-6A6968EBADED@freeswitch.org> I actually use Proxmox VE which provides a nice web ui for managing both OpenVZ and KVM instances. Proxmox VE is built on top of Debian Squeeze (not sure if they have started moving their dev to wheezy at this point) it works well. many services on the freeswitch.org domain are ran this way also. one of the things i use proxmox to accomplish is host managament. i'm running the entire box dedicated to 1 FS instance, but still use openvz so i can move an instance to a backup box incase of issues Ken Sent from my iPad On May 26, 2013, at 12:09, Tamas Jalsovszky wrote: > Hello Ken, > > Do you use any kind of virtualizaztion/separation on Debian? We would like to keep lighweight 'virtualization' with openvz or maybe lxc (or any other idea?). Any experience with that on Debian? > Is Debian better suited for FS than Ubuntu LTS (e.g. 12.04)? > It seems, we have to find out where to go from latest centos5+ovz+ubuntu 10.04 in VE... > > T. > > > On Sat, May 25, 2013 at 4:42 PM, Ken Rice wrote: >> I actually dropped using Centos6 and moved to debian to get both timerfd and to get performance at the same time. And things started working much nicer... Not sure if they even fixed the performance issues on centos tho >> >> >> >> On 5/25/13 4:52 AM, "Yuriy Nasida" wrote: >> >> Tamas, >> >> I think you can find info about performance problems with CentOS 6 on jira. >> For example. http://jira.freeswitch.org/browse/FS-4291 >> >> We also wanted to use timerfd (without virtualization) and made a moving to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen sometimes. As a result we had to move on centos 5.9 again. >> >> I would like to join issue. how can I be sure that timerfd is used? >> >> Regards, >> Yuriy >> >> Date: Sat, 25 May 2013 07:52:28 +0200 >> From: jalsot at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] OpenVZ tuning tips >> >> We've tried as root user but as I remember it was not able to set up the priority but will recheck to be sure. Another interesting thing is that when I tried to set a running process scheduler with chrt, got operation not permitted (as root of course), so I guess, something has to be tuned in the VE or on the host. We will try on bare metal centos6+ovz on the host. >> We try the latest centos6 with latest openvz kernel (due to security requirements we run on latest stable kernel and OS versions). >> Could you give some info about those horrible performance problems to let us check whether it still applies or not? (we've found only mysql create table performance degradation due to ext4 - where the solution could be barrier=0 yet, no other problems). >> Actually, how can I be sure that timerfd is used? strace? I'm nearly sure that timerfd works fine in FS. >> >> Yep, it would be much simpler without virtualization, and much harder from another perspective. Probably lxc, kvm and xen aren't much better regarding realtime stuff... >> >> Regards, >> Tamas >> >> >> On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale wrote: >> It will only work running as root I believe because it needs high privs to do realtime. >> If you do use centOS make sure its the latest rev of cent6, we have some horrible performance problems on the earlier revs and I don't know if they were resolved. >> Just make sure your kernels are as new as you can get them on the real host to avoid years of kernel performance bugs and that it has support for timerfd otherwise your VE would compile with timerfd support but not have actual access to the real syscalls for it in the host. >> >> That's why we try not to recommend virtual stuff in general as it takes some very careful setups and its hard to support from our standpoint when people run into issues. >> >> >> >> >> >> On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky wrote: >> Hello, >> >> Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on host and Ubuntu 10.04 LTS in VE. >> Do you know maybe how to allow realtime priority in the VE for FS? Running FS with -rp does not set the scheduler. >> strace says, sched_setscheduler operation permitted, so SCHED_FF is not set. Tried to run as root and/or use ulimit -r option, but cannot run FS with tuned priorities.We guess, some thing missing in the host/VE configuraton. >> Any idea? >> >> Br, >> Tamas >> >> >> On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: >> Im using 2.6.32 on all my boxes ... >> >> One thing that has me thinking, are there any tweaks to get MSI-X working best it can ? ( with proxmox ) >> there seems to be a strong bias towards one CPU for all interrupts. >> >> I could be wrong, but its something I think ive seen, and didnt see any clear suggestions on. >> >> Jay >> >> >> >> >> >> On 23 May 2013 01:12, Anthony Minessale wrote: >> 2.6.25 or newer to get timerfd support. >> >> >> >> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >> On 5/22/13, Anthony Minessale wrote: >> > You should consider centos6 or debian stable. Make sure the host kernel is >> > very new to get maximum results. >> >> Tony, do you mean "very new kernel" means 3.2.xx kernel? >> >> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >> possible. And that is what CentOS6 offers, too. >> >> However, I installed FS as openvz guest, it works fine for outgoing, >> but not DNAT works for incoming connections even after throroughly >> following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >> >> Just my two cents. >> >> >> >> > >> > >> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky wrote: >> > >> >> Hello, >> >> >> >> Do you have any recommendations regarding how to set up correctly (for >> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> >> tuneup >> >> the system to be rock solid? >> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS >> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> >> something is not well configured around timers, however mod_posix_timer >> >> did >> >> not help anything (running FS with -rp). We use our own bare metal and >> >> can >> >> reproduce those delatas eirher when only one VE is on the HW. >> >> Maybe time to check out centos6 with openvz? >> >> >> >> Any idea, recommendation, experience can be very helpful. >> >> >> >> Regards, >> >> Jalsot >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.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 >> >> >> >> >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> -- >> Ken >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/3065c609/attachment-0001.html From eoaddai at gmail.com Sun May 26 15:20:56 2013 From: eoaddai at gmail.com (eoaddai) Date: Sun, 26 May 2013 04:20:56 -0700 (PDT) Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION In-Reply-To: References: <1369468901841-7591099.post@n2.nabble.com> Message-ID: So, is the call hitting the goip at all? or, freeswitch is unable to process it to the goip? On 26 May 2013 10:32, Daniel Ivanov [via freeswitch-users] < ml-node+s2379917n7591120h93 at n2.nabble.com> wrote: > Maybe you're not sending them the right codecs or trying to run a feature > they don't have. Revisit your vars.xml and sip_profiles .xml. . Ultimately > contact the provider to ask them what youre doing wrong in your sdps. > On May 26, 2013 11:26 AM, "eoaddai" <[hidden email]> > wrote: > >> Hi, my first time of posting stuff here. I really need help. >> I keep on not getting my calls go through freeswitch to my goip gateway. >> The >> following is the freeswitch log with siptrace turned on i get. Please help >> me: >> >> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing >> global variables >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [origination_caller_id_number]=[0264370536] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_from]=[0264370536] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_to]=[0267577771] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_app]=[true] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [originate_timeout]=[60] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >> [ignore_early_media]=[true] >> 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel >> sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] >> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 >> (sofia/external/0267577771) State Change CS_NEW -> CS_INIT >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0267577771) Running State Change CS_INIT >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/0267577771) State INIT >> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 >> sofia/external/0267577771 >> SOFIA INIT >> 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: >> v=0 >> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >> s=FreeSWITCH >> c=IN IP4 10.10.50.1 >> t=0 0 >> m=audio 30174 RTP/AVP 0 8 3 101 13 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 >> (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/0267577771) State INIT going to sleep >> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: >> >> ------------------------------------------------------------------------ >> INVITE [hidden email]SIP/2.0 >> >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >> Max-Forwards: 70 >> From: "" <[hidden email] >> >;tag=DXjN0tK6mcc5S >> To: <[hidden email] >> > >> >> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, hold, conference, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 201 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: >> <[hidden email] >> >;party=calling;screen=yes;privacy=off >> >> >> v=0 >> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >> s=FreeSWITCH >> c=IN IP4 10.10.50.1 >> t=0 0 >> m=audio 30174 RTP/AVP 0 8 3 101 13 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0267577771) Running State Change CS_ROUTING >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/0267577771) State ROUTING >> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 >> sofia/external/0267577771 >> SOFIA ROUTING >> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 >> (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/0267577771) State ROUTING going to sleep >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >> (sofia/external/0267577771) State CONSUME_MEDIA >> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >> (sofia/external/0267577771) State CONSUME_MEDIA going to sleep >> 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel >> sofia/external/0267577771 entering state [calling][0] >> recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: >> >> ------------------------------------------------------------------------ >> SIP/2.0 606 Not Acceptable >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >> From: "" <[hidden email] >> >;tag=DXjN0tK6mcc5S >> To: <[hidden email] >> >;tag=1662509363 >> >> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 INVITE >> User-Agent: dble >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: >> >> ------------------------------------------------------------------------ >> ACK [hidden email]SIP/2.0 >> >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >> Max-Forwards: 70 >> From: "" <[hidden email] >> >;tag=DXjN0tK6mcc5S >> To: <[hidden email] >> >;tag=1662509363 >> >> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 ACK >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel >> sofia/external/0267577771 entering state [terminated][606] >> 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup >> sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 Originate >> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal >> sofia/external/0267577771 [KILL] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0267577771) Running State Change CS_HANGUP >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/0267577771) State HANGUP >> 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel >> sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/0267577771 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/0267577771) State HANGUP going to sleep >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 >> (sofia/external/0267577771) Callstate Change DOWN -> HANGUP >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 >> (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0267577771) Running State Change CS_REPORTING >> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/0267577771) State REPORTING >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/0267577771 Standard REPORTING, cause: >> INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/0267577771) State REPORTING going to sleep >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0267577771 [BREAK] >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 >> (sofia/external/0267577771) Locked, Waiting on external entities >> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session 3 >> (sofia/external/0267577771) Ended >> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close >> Channel >> sofia/external/0267577771 [CS_DESTROY] >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 >> (sofia/external/0267577771) Callstate Change HANGUP -> DOWN >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 >> (sofia/external/0267577771) Running State Change CS_DESTROY >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >> (sofia/external/0267577771) State DESTROY >> 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 >> sofia/external/0267577771 >> SOFIA DESTROY >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 >> sofia/external/0267577771 Standard DESTROY >> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >> (sofia/external/0267577771) State DESTROY going to sleep >> 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing >> global variables >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [origination_caller_id_number]=[0264370536] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_from]=[0264370536] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_to]=[0249230704] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [plivo_app]=[true] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [originate_timeout]=[60] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >> [ignore_early_media]=[true] >> 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel >> sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] >> 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 >> (sofia/external/0249230704) State Change CS_NEW -> CS_INIT >> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0249230704) Running State Change CS_INIT >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/0249230704) State INIT >> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 >> sofia/external/0249230704 >> SOFIA INIT >> 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: >> v=0 >> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >> s=FreeSWITCH >> c=IN IP4 10.10.50.1 >> t=0 0 >> m=audio 19250 RTP/AVP 0 8 3 101 13 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 >> (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >> (sofia/external/0249230704) State INIT going to sleep >> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: >> >> ------------------------------------------------------------------------ >> INVITE [hidden email]SIP/2.0 >> >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >> Max-Forwards: 70 >> From: "" <[hidden email] >> >;tag=e6Be2N49HN2QN >> To: <[hidden email] >> > >> >> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 INVITE >> Contact: >> User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, hold, conference, refer >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 201 >> X-FS-Support: update_display,send_info >> Remote-Party-ID: >> <[hidden email] >> >;party=calling;screen=yes;privacy=off >> >> >> v=0 >> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >> s=FreeSWITCH >> c=IN IP4 10.10.50.1 >> t=0 0 >> m=audio 19250 RTP/AVP 0 8 3 101 13 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> >> ------------------------------------------------------------------------ >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0249230704) Running State Change CS_ROUTING >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/0249230704) State ROUTING >> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 >> sofia/external/0249230704 >> SOFIA ROUTING >> 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 >> (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/0249230704) State ROUTING going to sleep >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >> (sofia/external/0249230704) State CONSUME_MEDIA >> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >> (sofia/external/0249230704) State CONSUME_MEDIA going to sleep >> 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel >> sofia/external/0249230704 entering state [calling][0] >> recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >> >> ------------------------------------------------------------------------ >> SIP/2.0 606 Not Acceptable >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >> From: "" <[hidden email] >> >;tag=e6Be2N49HN2QN >> To: <[hidden email] >> >;tag=633680086 >> >> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 INVITE >> User-Agent: dble >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: >> >> ------------------------------------------------------------------------ >> ACK [hidden email]SIP/2.0 >> >> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >> Max-Forwards: 70 >> From: "" <[hidden email] >> >;tag=e6Be2N49HN2QN >> To: <[hidden email] >> >;tag=633680086 >> >> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >> CSeq: 44398096 ACK >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel >> sofia/external/0249230704 entering state [terminated][606] >> 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup >> sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal >> sofia/external/0249230704 [KILL] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0249230704) Running State Change CS_HANGUP >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/0249230704) State HANGUP >> 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel >> sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/0249230704 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/0249230704) State HANGUP going to sleep >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 >> (sofia/external/0249230704) Callstate Change DOWN -> HANGUP >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 >> (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/0249230704) Running State Change CS_REPORTING >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/0249230704) State REPORTING >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/0249230704 Standard REPORTING, cause: >> INCOMPATIBLE_DESTINATION >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/0249230704) State REPORTING going to sleep >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >> sofia/external/0249230704 [BREAK] >> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 >> (sofia/external/0249230704) Locked, Waiting on external entities >> 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 Originate >> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session 4 >> (sofia/external/0249230704) Ended >> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close >> Channel >> sofia/external/0249230704 [CS_DESTROY] >> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 >> (sofia/external/0249230704) Callstate Change HANGUP -> DOWN >> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 >> (sofia/external/0249230704) Running State Change CS_DESTROY >> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >> (sofia/external/0249230704) State DESTROY >> 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 >> sofia/external/0249230704 >> SOFIA DESTROY >> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 >> sofia/external/0249230704 Standard DESTROY >> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >> (sofia/external/0249230704) State DESTROY going to sleep >> >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> >> 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: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591120.html > To unsubscribe from SIP/2.0 606 Not Acceptable and > INCOMPATIBLE_DESTINATION, click here > . > NAML > -- *Emmanuel O. Addai,* *+233(0)26 757 7771* -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591122.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130526/eac7ab6b/attachment-0001.html From adam at adamcooley.com Mon May 27 07:16:21 2013 From: adam at adamcooley.com (Adam Cooley) Date: Sun, 26 May 2013 22:16:21 -0500 Subject: [Freeswitch-users] add-users scripts Message-ID: hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via centos, with yum, and was wondering where I would procure the add-user scripts for adding users on the cli? i can't seem to find them on the filesystem.. -adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130526/077136cb/attachment.html From bdfoster at davri.com Mon May 27 07:57:04 2013 From: bdfoster at davri.com (Brian Foster) Date: Sun, 26 May 2013 23:57:04 -0400 Subject: [Freeswitch-users] add-users scripts In-Reply-To: References: Message-ID: Users are added via XML by default. The files are located in /usr/local/freeswitch/conf/directory/default. There are a few files in there to give yiu an idea of what users look like. Afterwards you should load up fs_cli and issue 'reloadxml' without quotes to put your changes into effect. You should thoroughly read and search through the project's wiki page; it's full of extremely valuable information. You can find it here: http://wiki.freeswitch.org. For more information regarding Users in the directory, search for 'XML User Directory' on the wiki. Welcome to the FreeSWITCH community! We're a very active community and can be found primarily on IRC (#freeswitch on freenode), here on the users mailing list, and every Wednesday around 1pm EST for a conference call. We're here to help! - BDF On May 26, 2013 11:46 PM, "Adam Cooley" wrote: > hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via centos, > with yum, and was wondering where I would procure the add-user scripts for > adding users on the cli? i can't seem to find them on the filesystem.. > > -adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/daf69b0c/attachment.html From sertys at gmail.com Mon May 27 09:14:13 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Mon, 27 May 2013 07:14:13 +0200 Subject: [Freeswitch-users] add-users scripts In-Reply-To: References: Message-ID: You can use this ruby script if you want. https://github.com/lazzarello/chef-twelvetone/blob/master/cookbooks/freeswitch/templates/default/gen_users.rb.erb Part of tom azarello's ostn recipes On May 27, 2013 6:43 AM, "Adam Cooley" wrote: > hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via centos, > with yum, and was wondering where I would procure the add-user scripts for > adding users on the cli? i can't seem to find them on the filesystem.. > > -adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/f6fb0e28/attachment.html From ashish at nms.co.in Mon May 27 09:49:02 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 27 May 2013 11:19:02 +0530 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: <519F6087.3030706@mst.edu> References: <519F5CCD.8000609@mst.edu> <519F6087.3030706@mst.edu> Message-ID: Hi Nathan, Even setting the api_hangup_hook=perl hook.pl in the originate string does not work. hook.pl does not get executed on hangup. It has to be done some other way I guess. Thanks. On Fri, May 24, 2013 at 6:13 PM, Nathan Neulinger wrote: > I don't think that's going to do what you want... (May be wrong.) > > I think that continue_on_fail is only going to apply to the rules for the > received call on this extension, not the received call on the outgoing leg. > > i.e. there are no dialplan rules in effect for the outgoing call that you > initiated, and that's where the failure is occurring. For these dialplan > rules, I think the only failure would be if your IVR (I assume that's was > ash.pl is) didn't answer. > > Like I said, not certain of this, maybe some else can chime in, but I > think you're going to have to handle that failure as a part of your > originate on the outbound call. Something like putting > > originate {api_hangup_hook=perl hook.pl}sofia/..... > > Where you cause the call to take place. > > -- Nathan > > > On 05/24/2013 07:37 AM, Ashish gautam wrote: > >> I am generating an outgoing call through mod_event_socket and then >> transferring it to this dialplan. >> >> On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger > nneul at mst.edu>> wrote: >> >> I may be misunderstanding - but where are you causing it to ring a >> device? >> >> You've told it to internally answer the call, and then not do >> anything. There's no bridging to an actual extension. >> >> Only thing I see that would happen is it running perl/ash.pl < >> http://ash.pl>, unclear if it would in term execute >> hook.pl when that script finished (I don't know >> what that behavior is expected to be). >> >> >> -- Nathan >> >> >> On 05/24/2013 07:17 AM, Ashish gautam wrote: >> >> Hi, >> >> I have a dialplan as follows: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> when the called party does not pick up the phone or is busy, the >> dialplan does not proceed and hook.pl >> >> >> >> does not get executed. >> >> Please help >> -- >> Ashish Gautam >> >> IVR Developer >> >> Nucleus Microsystems (Pvt.) Ltd. >> >> >> >> -- >> ------------------------------**__----------------------------**-- >> Nathan Neulinger nneul at mst.edu >> >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> >> >> >> >> -- >> Ashish Gautam >> >> IVR Developer >> >> Nucleus Microsystems (Pvt.) Ltd. >> >> Ph. 011 47574758 >> > > -- > ------------------------------**------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd. Ph. 011 47574758 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/0d1da3e1/attachment.html From adam at adamcooley.com Mon May 27 08:55:40 2013 From: adam at adamcooley.com (Adam Cooley) Date: Sun, 26 May 2013 23:55:40 -0500 Subject: [Freeswitch-users] add-users scripts In-Reply-To: References: Message-ID: Thanks brian, I'm just wondering where I get the add_users script referenced in the freeswitch cookbook and others? Also finding with a fresh install of freeswitch 1.2.9 with centos and yum, non of the .wav files are loaded for voicemail. Where do I get those files from? -adam On Sun, May 26, 2013 at 10:57 PM, Brian Foster wrote: > Users are added via XML by default. The files are located in > /usr/local/freeswitch/conf/directory/default. There are a few files in > there to give yiu an idea of what users look like. Afterwards you should > load up fs_cli and issue 'reloadxml' without quotes to put your changes > into effect. > > You should thoroughly read and search through the project's wiki page; > it's full of extremely valuable information. You can find it here: > http://wiki.freeswitch.org. For more information regarding Users in the > directory, search for 'XML User Directory' on the wiki. > > Welcome to the FreeSWITCH community! We're a very active community and can > be found primarily on IRC (#freeswitch on freenode), here on the users > mailing list, and every Wednesday around 1pm EST for a conference call. > We're here to help! > > - BDF > On May 26, 2013 11:46 PM, "Adam Cooley" wrote: > >> hey gang, new to freeswitch here..I installed freeswitch 1.2.9 via >> centos, with yum, and was wondering where I would procure the add-user >> scripts for adding users on the cli? i can't seem to find them on the >> filesystem.. >> >> -adam >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130526/adba45f4/attachment-0001.html From adam at adamcooley.com Mon May 27 09:21:56 2013 From: adam at adamcooley.com (Adam Cooley) Date: Mon, 27 May 2013 00:21:56 -0500 Subject: [Freeswitch-users] missing wav files from yum install? Message-ID: Got freeswitch installed, and debugs on fs_cli show were missing audio files with a fresh install of 1.2.9 from yum on centos. 2013-05-25 19:48:27.071848 [ERR] mod_sndfile.c:202 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/ivr/ivr-to_call_the_freeswitch_conference.wav] [System error : No such file or directory.] Where do we get the default system wav files from? -a -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/2cd76c40/attachment.html From ashish at nms.co.in Mon May 27 12:56:04 2013 From: ashish at nms.co.in (Ashish gautam) Date: Mon, 27 May 2013 14:26:04 +0530 Subject: [Freeswitch-users] Freetdm PRI Span Failover Message-ID: Hi, I have been using single span PRI card with FreeTDM but now I plan to use 8 Span card. I want to know if the channels on a particular span are all busy(or unavailable) at a time and a call needs to be originated, then how to automatically pass it through the other free span (failover kind of a thing) on which the channels are available? Please help! -- Ashish Gautam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/9bcb2562/attachment.html From smfarrukh at live.com Mon May 27 14:35:26 2013 From: smfarrukh at live.com (Farrukh Ali) Date: Mon, 27 May 2013 10:35:26 +0000 Subject: [Freeswitch-users] Anyone used Instant messaging with FS Message-ID: Dear all, I am trying to configure and use instant messaging with FS, can anyone give me a hint about configuration in FS and the soft phones? Regards, Muhammad Farrukh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/7057e5ae/attachment.html From jalsot at gmail.com Mon May 27 14:51:21 2013 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Mon, 27 May 2013 12:51:21 +0200 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: <9BCB536D-E68D-4AD7-80CD-6A6968EBADED@freeswitch.org> References: <9BCB536D-E68D-4AD7-80CD-6A6968EBADED@freeswitch.org> Message-ID: Hello Ken, As I've checked, there is new proxmox version based on Wheezy. As I've found, Proxmox kernel is based on RHEL6x, so probably it needs intensive testing too. What were the reasons you decided to move off from CentOS? Were you able to reproduce problems? Br, Tamas On Sun, May 26, 2013 at 8:12 PM, Ken Rice wrote: > I actually use Proxmox VE which provides a nice web ui for managing both > OpenVZ and KVM instances. Proxmox VE is built on top of Debian Squeeze (not > sure if they have started moving their dev to wheezy at this point) > > it works well. many services on the freeswitch.org domain are ran this > way also. > > one of the things i use proxmox to accomplish is host managament. i'm > running the entire box dedicated to 1 FS instance, but still use openvz so > i can move an instance to a backup box incase of issues > > > > Ken > Sent from my iPad > > On May 26, 2013, at 12:09, Tamas Jalsovszky wrote: > > Hello Ken, > > Do you use any kind of virtualizaztion/separation on Debian? We would like > to keep lighweight 'virtualization' with openvz or maybe lxc (or any other > idea?). Any experience with that on Debian? > Is Debian better suited for FS than Ubuntu LTS (e.g. 12.04)? > It seems, we have to find out where to go from latest centos5+ovz+ubuntu > 10.04 in VE... > > T. > > > On Sat, May 25, 2013 at 4:42 PM, Ken Rice wrote: > >> I actually dropped using Centos6 and moved to debian to get both >> timerfd and to get performance at the same time. And things started working >> much nicer... Not sure if they even fixed the performance issues on centos >> tho >> >> >> >> On 5/25/13 4:52 AM, "Yuriy Nasida" wrote: >> >> Tamas, >> >> I think you can find info about performance problems with CentOS 6 on >> jira. >> For example. http://jira.freeswitch.org/browse/FS-4291 >> >> We also wanted to use timerfd (without virtualization) and made a moving >> to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen >> sometimes. As a result we had to move on centos 5.9 again. >> >> I would like to join issue. how can I be sure that timerfd is used? >> >> Regards, >> Yuriy >> >> ------------------------------ >> Date: Sat, 25 May 2013 07:52:28 +0200 >> From: jalsot at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] OpenVZ tuning tips >> >> We've tried as root user but as I remember it was not able to set up the >> priority but will recheck to be sure. Another interesting thing is that >> when I tried to set a running process scheduler with chrt, got operation >> not permitted (as root of course), so I guess, something has to be tuned in >> the VE or on the host. We will try on bare metal centos6+ovz on the host. >> We try the latest centos6 with latest openvz kernel (due to security >> requirements we run on latest stable kernel and OS versions). >> Could you give some info about those horrible performance problems to let >> us check whether it still applies or not? (we've found only mysql create >> table performance degradation due to ext4 - where the solution could be >> barrier=0 yet, no other problems). >> Actually, how can I be sure that timerfd is used? strace? I'm nearly sure >> that timerfd works fine in FS. >> >> Yep, it would be much simpler without virtualization, and much harder >> from another perspective. Probably lxc, kvm and xen aren't much better >> regarding realtime stuff... >> >> Regards, >> Tamas >> >> >> On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >> It will only work running as root I believe because it needs high privs >> to do realtime. >> If you do use centOS make sure its the latest rev of cent6, we have some >> horrible performance problems on the earlier revs and I don't know if they >> were resolved. >> Just make sure your kernels are as new as you can get them on the real >> host to avoid years of kernel performance bugs and that it has support for >> timerfd otherwise your VE would compile with timerfd support but not have >> actual access to the real syscalls for it in the host. >> >> That's why we try not to recommend virtual stuff in general as it takes >> some very careful setups and its hard to support from our standpoint when >> people run into issues. >> >> >> >> >> >> On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky >> wrote: >> >> Hello, >> >> Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on >> host and Ubuntu 10.04 LTS in VE. >> Do you know maybe how to allow realtime priority in the VE for FS? >> Running FS with -rp does not set the scheduler. >> strace says, sched_setscheduler operation permitted, so SCHED_FF is not >> set. Tried to run as root and/or use ulimit -r option, but cannot run FS >> with tuned priorities.We guess, some thing missing in the host/VE >> configuraton. >> Any idea? >> >> Br, >> Tamas >> >> >> On Thu, May 23, 2013 at 12:47 AM, jay binks wrote: >> >> Im using 2.6.32 on all my boxes ... >> >> One thing that has me thinking, are there any tweaks to get MSI-X >> working best it can ? ( with proxmox ) >> there seems to be a strong bias towards one CPU for all interrupts. >> >> I could be wrong, but its something I think ive seen, and didnt see any >> clear suggestions on. >> >> Jay >> >> >> >> >> >> On 23 May 2013 01:12, Anthony Minessale >> wrote: >> >> 2.6.25 or newer to get timerfd support. >> >> >> >> On Wed, May 22, 2013 at 2:56 AM, Zenny wrote: >> >> On 5/22/13, Anthony Minessale wrote: >> > You should consider centos6 or debian stable. Make sure the host >> kernel is >> > very new to get maximum results. >> >> Tony, do you mean "very new kernel" means 3.2.xx kernel? >> >> Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not >> possible. And that is what CentOS6 offers, too. >> >> However, I installed FS as openvz guest, it works fine for outgoing, >> but not DNAT works for incoming connections even after throroughly >> following >> http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. >> >> Just my two cents. >> >> >> >> > >> > >> > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky >> wrote: >> > >> >> Hello, >> >> >> >> Do you have any recommendations regarding how to set up correctly (for >> >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> >> tuneup >> >> the system to be rock solid? >> >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with >> FS >> >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> >> something is not well configured around timers, however mod_posix_timer >> >> did >> >> not help anything (running FS with -rp). We use our own bare metal and >> >> can >> >> reproduce those delatas eirher when only one VE is on the HW. >> >> Maybe time to check out centos6 with openvz? >> >> >> >> Any idea, recommendation, experience can be very helpful. >> >> >> >> Regards, >> >> Jalsot >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.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 >> >> >> >> >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com < >> mailto:MSN%3Aanthony_minessale at hotmail.com> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com < >> mailto:PAYPAL%3Aanthony.minessale at gmail.com> >> >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org < >> mailto:sip%3A888 at conference.freeswitch.org> >> >> > googletalk:conf+888 at conference.freeswitch.org < >> mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> >> >> > pstn:+19193869900 >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> Ken >> *http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> *irc.freenode.net #freeswitch >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/4cfbb7b9/attachment-0001.html From sertys at gmail.com Mon May 27 17:29:51 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Mon, 27 May 2013 15:29:51 +0200 Subject: [Freeswitch-users] Anyone used Instant messaging with FS In-Reply-To: References: Message-ID: There're threads in the archives about it. Look them up, please. On Mon, May 27, 2013 at 12:35 PM, Farrukh Ali wrote: > Dear all, > > I am trying to configure and use instant messaging with FS, can anyone > give me a hint about configuration in FS and the soft phones? > > > Regards, > Muhammad Farrukh > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/283459e2/attachment.html From sertys at gmail.com Mon May 27 17:53:11 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Mon, 27 May 2013 15:53:11 +0200 Subject: [Freeswitch-users] Freeswitch just won't transcode the call. Message-ID: I'm having media problems with a FS here. UAs are sending mixed capabilites all the time and i wanna gracefully degradate on them. This is what i have in vars.xml: This is in external.xml: I need to make sure that when a UA dials in, the A-leg is negotiated between g729 and gsm(and codec2) and the call is then transcoded to the B-leg if it's not G729 instead of just resending the SDP directly to the gateways. I've tried disabling and enabling proxy-media, late-negotiation and disable-transcoding along with passing the strict codec_string and absolute_codec_string. Point me on what i'm failing to apprehend here. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/1d274b3f/attachment.html From tculjaga at gmail.com Mon May 27 18:13:44 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 27 May 2013 16:13:44 +0200 Subject: [Freeswitch-users] bridging 2 call legs after a conference Message-ID: hello folks, im controlling FS via ESL and what im trying to do is to make a 3-way conference and this works fine. What i have an issue with is when one conference member hangs up, i want to return back to a normal bridge between the 2 remaining members ... so what im doing is: On Incoming call from 1013 to FS answer the call: ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: answer"]) play an announcement to A leg: CMD uuid_broadcast "bgapi uuid_broadcast 00349e91-117e-4dcc-b2db-45d3a54c5fee 'greeting1.wav'" originate a new call and parks it: CMD originate "bgapi originate {origination_uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234,ignore_early_media=false,originate_timeout=15}user/1002 &park" bridge these 2 legs into a call: CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" originate a 2nd new call into a conference: CMD originate "bgapi originate {origination_uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f,ignore_early_media=false,originate_timeout=0}user/1014 &conference(supervizor36 at silent+flags{mute}) &park" join the 1st new call (user/1002) CMD_MULTI conference ["sendmsg fc3e44c5-63af-45bb-8a8e-8a35bac52234","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36 at silent"] join incoming call into the same conference CMD_MULTI conference ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36 at silent"] unmute uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 unmute 20" Incoming call 00349e91-117e-4dcc-b2db-45d3a54c5fee must not hear uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 relate 21 20 nohear" so far its great! now, call uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f (the one originated into the conference) hangs up, 2 remaining calls (incoming call and 1st originated calls) can still hear each others... but i want to get rid of a conference since there are just 2 members hence no sense keeping the conference up... so i do: bridge the two remaining conference members together CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" and i get no audio!! looks like FS setting both channels on park ?? any idea ? 1. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1367 Session 30 (sofia/internal/sip:1014 at 192.168.254.112:5060) Ended 2. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:1014 at 192.168.254.112:5060 [CS_DESTROY] 3. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:491 ( sofia/internal/sip:1014 at 192.168.254.112:5060) Callstate Change HANGUP -> DOWN 4. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:494 ( sofia/internal/sip:1014 at 192.168.254.112:5060) Running State Change CS_DESTROY 5. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY 6. 2013-05-27 15:34:13.724037 [DEBUG] mod_sofia.c:363 sofia/internal/sip: 1014 at 192.168.254.112:5060 SOFIA DESTROY 7. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1014 at 192.168.254.112:5060 Standard DESTROY 8. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY going to sleep 9. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1521 (sofia/internal/ sip:1002 at 192.168.254.116:5060) State Change CS_SOFT_EXECUTE -> CS_HIBERNATE 10. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 11. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1523 ( sofia/external/1013 at 192.168.254.93) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE 12. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 13. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 14. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 15. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE 16. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE 17. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 18. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/internal/sip:1002 at 192.168.254.116:5060 Restore previous codec PCMA:8. 19. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 20. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 21. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 22. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/external/1013 at 192.168.254.93 Restore previous codec PCMA:8. 23. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 24. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 25. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1602 Write Lock ON 26. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1002 at 192.168.254.116:5060] 27. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1605 Write Lock OFF 28. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 29. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 30. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 31. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:1366 ( sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_HIBERNATE -> CS_RESET 32. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 33. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/ sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE going to sleep 34. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change CS_RESET 35. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 ( sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET 36. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1002 at 192.168.254.116:5060 SOFIA RESET 37. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:721 sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM RESET 38. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:728 ( sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_RESET -> CS_SOFT_EXECUTE 39. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 40. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 ( sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET going to sleep 41. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change CS_SOFT_EXECUTE 42. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 ( sofia/internal/sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE 43. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:572 SOFIA SOFT_EXECUTE 44. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:746 sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM SOFT_EXECUTE 45. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/1013 at 192.168.254.93] 46. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] 47. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr.c:2422 (sofia/external/ 1013 at 192.168.254.93) State Change CS_HIBERNATE -> CS_PARK 48. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] 49. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:420 ( sofia/external/1013 at 192.168.254.93) State EXCHANGE_MEDIA going to sleep 50. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/external/1013 at 192.168.254.93) Running State Change CS_PARK 51. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:426 ( sofia/external/1013 at 192.168.254.93) State PARK 52. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:247 sofia/external/1013 at 192.168.254.93 Standard PARK 53. 54. 55. 56. freeswitch at internal> 2013-05-27 15:34:18.457165 [WARNING] sofia_reg.c:1400 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1014 at 192.168.254.93]from ip 192.168.254.112 57. 58. freeswitch at internal> 59. freeswitch at internal> 60. freeswitch at internal> show calls 61. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 62. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168 .254.112,38515494471,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,cc01,Outbound Call,1002,,,,,,,,,,,,,,,,,,, 63. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,1002 at 192.168.254.93 ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,cc01,1013,1013 ,,,,,,,,,,,,,,,,,,, 64. 65. 2 total. 66. 67. freeswitch at internal> show channels 68. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 69. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168 .254.112,38515494471,conference,supervizor36 at silent,XML,default,PCMA,8000 ,64000,PCMA,8000,64000,,cc01,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,Outbound Call,1002 70. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,conference,supervizor36 at silent,,default,PCMA,8000,64000 ,PCMA,8000,64000,,cc01,1002 at 192.168.254.93 ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,1013,1013 71. 72. 2 total. 73. 74. freeswitch at internal> 75. freeswitch at internal> 76. freeswitch at internal> 77. freeswitch at internal> 2013-05-27 15:34:57.732290 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1002 at 192.168 .254.116:5060 [BREAK] Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/b87f1cca/attachment-0001.html From tculjaga at gmail.com Mon May 27 19:21:02 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 27 May 2013 17:21:02 +0200 Subject: [Freeswitch-users] bridging 2 call legs after a conference In-Reply-To: References: Message-ID: ya, i have park_after_bridge=true on the incoming call leg :=) ... thats why it goes on park after leaving the conference... never thought park_after_bridge affects conference as well ... :) thanks anyway. T. On Mon, May 27, 2013 at 4:13 PM, Tihomir Culjaga wrote: > hello folks, > > > im controlling FS via ESL and what im trying to do is to make a 3-way > conference and this works fine. What i have an issue with is when one > conference member hangs up, i want to return back to a normal bridge > between the 2 remaining members ... > > so what im doing is: > > On Incoming call from 1013 to FS > answer the call: > ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: > execute","execute-app-name: answer"]) > > play an announcement to A leg: > CMD uuid_broadcast "bgapi uuid_broadcast > 00349e91-117e-4dcc-b2db-45d3a54c5fee 'greeting1.wav'" > > > originate a new call and parks it: > CMD originate "bgapi originate > {origination_uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234,ignore_early_media=false,originate_timeout=15}user/1002 > &park" > > > bridge these 2 legs into a call: > CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 > 00349e91-117e-4dcc-b2db-45d3a54c5fee" > > > > originate a 2nd new call into a conference: > CMD originate "bgapi originate > {origination_uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f,ignore_early_media=false,originate_timeout=0}user/1014 > &conference(supervizor36 at silent+flags{mute}) &park" > > > join the 1st new call (user/1002) > CMD_MULTI conference ["sendmsg > fc3e44c5-63af-45bb-8a8e-8a35bac52234","call-command: > execute","execute-app-name: conference","execute-app-arg: > supervizor36 at silent"] > > join incoming call into the same conference > CMD_MULTI conference ["sendmsg > 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: > execute","execute-app-name: conference","execute-app-arg: > supervizor36 at silent"] > > > unmute uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 > CMD api "api conference supervizor36 unmute 20" > > > Incoming call 00349e91-117e-4dcc-b2db-45d3a54c5fee must not hear > uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 > CMD api "api conference supervizor36 relate 21 20 nohear" > > > so far its great! > > now, call uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f (the one originated > into the conference) hangs up, 2 remaining calls (incoming call and 1st > originated calls) can still hear each others... but i want to get rid of a > conference since there are just 2 members hence no sense keeping the > conference up... > > so i do: > bridge the two remaining conference members together > CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 > 00349e91-117e-4dcc-b2db-45d3a54c5fee" > > and i get no audio!! > > > > looks like FS setting both channels on park ?? > > any idea ? > > > > 1. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1367 > Session 30 (sofia/internal/sip:1014 at 192.168.254.112:5060) Ended > 2. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1369 Close > Channel sofia/internal/sip:1014 at 192.168.254.112:5060 [CS_DESTROY] > 3. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:491 ( > sofia/internal/sip:1014 at 192.168.254.112:5060) Callstate Change HANGUP > -> DOWN > 4. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:494 ( > sofia/internal/sip:1014 at 192.168.254.112:5060) Running State Change > CS_DESTROY > 5. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY > 6. 2013-05-27 15:34:13.724037 [DEBUG] mod_sofia.c:363 > sofia/internal/sip:1014 at 192.168.254.112:5060 SOFIA DESTROY > 7. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/sip:1014 at 192.168.254.112:5060 Standard DESTROY > 8. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( > sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY going to > sleep > 9. > 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1521 (sofia/internal/ > sip:1002 at 192.168.254.116:5060) State Change CS_SOFT_EXECUTE -> > CS_HIBERNATE > 10. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 11. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1523 ( > sofia/external/1013 at 192.168.254.93) State Change CS_EXCHANGE_MEDIA -> > CS_HIBERNATE > 12. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 13. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 14. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 15. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel > leaving conference, cause: NONE > 16. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel > leaving conference, cause: NONE > 17. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 18. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 > sofia/internal/sip:1002 at 192.168.254.116:5060 Restore previous codec > PCMA:8. > 19. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 20. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 21. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 22. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 > sofia/external/1013 at 192.168.254.93 Restore previous codec PCMA:8. > 23. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 24. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 25. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1602 Write > Lock ON > 26. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/internal/sip:1002 at 192.168.254.116:5060] > 27. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1605 Write > Lock OFF > 28. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 29. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 30. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 31. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:1366 ( > sofia/internal/sip:1002 at 192.168.254.116:5060) State Change > CS_HIBERNATE -> CS_RESET > 32. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 33. > 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/ > sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE going to sleep > 34. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change > CS_RESET > 35. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 > (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET > 36. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:166 > sofia/internal/sip:1002 at 192.168.254.116:5060 SOFIA RESET > 37. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:721 > sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM RESET > 38. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:728 ( > sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_RESET -> > CS_SOFT_EXECUTE > 39. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 40. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 > (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET going to > sleep > 41. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change > CS_SOFT_EXECUTE > 42. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 > (sofia/internal/sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE > 43. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:572 SOFIA > SOFT_EXECUTE > 44. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:746 > sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM SOFT_EXECUTE > 45. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE > THREAD DONE [sofia/external/1013 at 192.168.254.93] > 46. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send > signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] > 47. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr.c:2422 ( > sofia/external/1013 at 192.168.254.93) State Change CS_HIBERNATE -> > CS_PARK > 48. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send > signal sofia/external/1013 at 192.168.254.93 [BREAK] > 49. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:420 > (sofia/external/1013 at 192.168.254.93) State EXCHANGE_MEDIA going to > sleep > 50. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/1013 at 192.168.254.93) Running State Change CS_PARK > 51. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:426 > (sofia/external/1013 at 192.168.254.93) State PARK > 52. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:247 > sofia/external/1013 at 192.168.254.93 Standard PARK > 53. > 54. > 55. > 56. freeswitch at internal> 2013-05-27 15:34:18.457165 [WARNING] > sofia_reg.c:1400 SIP auth challenge (REGISTER) on sofia profile > 'internal' for [1014 at 192.168.254.93]from ip 192.168.254.112 > 57. > 58. freeswitch at internal> > 59. freeswitch at internal> > 60. freeswitch at internal> show calls > 61. > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch > 62. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, > 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168 > .254.112,38515494471,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,cc01,Outbound > Call,1002,,,,,,,,,,,,,,,,,,, > 63. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, > 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound > Call,1002,,1002,1002 at 192.168.254.93 > ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,cc01,1013,1013 > ,,,,,,,,,,,,,,,,,,, > 64. > 65. 2 total. > 66. > 67. freeswitch at internal> show channels > 68. > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num > 69. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, > 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168 > .254.112,38515494471,conference,supervizor36 at silent,XML,default,PCMA, > 8000,64000,PCMA,8000,64000,,cc01,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,Outbound > Call,1002 > 70. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, > 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound > Call,1002,,1002,conference,supervizor36 at silent,,default,PCMA,8000,64000 > ,PCMA,8000,64000,,cc01,1002 at 192.168.254.93 > ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,1013,1013 > 71. > 72. 2 total. > 73. > 74. freeswitch at internal> > 75. freeswitch at internal> > 76. freeswitch at internal> > 77. freeswitch at internal> 2013-05-27 15:34:57.732290 [DEBUG] > switch_core_session.c:870 Send signal sofia/internal/sip:1002 at 192.168 > .254.116:5060 [BREAK] > > > > > > Tihomir. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/fdba9b25/attachment-0001.html From jos at firstcom.dk Mon May 27 20:01:49 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Mon, 27 May 2013 18:01:49 +0200 Subject: [Freeswitch-users] Splitting CDRs on call forward Message-ID: Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Sch?pzinsky From krice at freeswitch.org Mon May 27 20:27:59 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 27 May 2013 11:27:59 -0500 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: Message-ID: if you are lusing xml cdr, set the logging to both not just a or b legs, this will generate cdrs for all legs individually Ken Sent from my iPad On May 27, 2013, at 11:01, Jon Sch?pzinsky wrote: > Hi List, > > I am implementing call forwarding on a multi tenant system, and therefore > need to split CDR's when the call forward happens, so that if the > receiving user also has his account call forwarded, he pays for his part > of the call. > > A calls B > B forwards to C > C forwards to an external mobile phone. > > B has a free call from B to C, but C needs to pay for the forwarding to > the mobile phone. Therefore i need a separate CDR for the C to Mobile > phone call. > > Another example would be this > > A works in Company A, and B works in Company B > They are both users on our system, and therefore is on the same freeswitch. > > A calls B > B Forwards to an external mobile phone. > > Here A needs to pay for the call from A to B, and B needs to pay for the > call being forwarded to his mobile phone. > > Do anybody have an idea as to how to implement this in freeswitch. Back in > my Asterisk days, this would be done by the ForkCDR command. > > > Venlig hilsen/kind regards > > Jon Leren Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From drk at drkngs.net Mon May 27 20:31:56 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Mon, 27 May 2013 09:31:56 -0700 Subject: [Freeswitch-users] bridging 2 call legs after a conference In-Reply-To: Message-ID: <20130527163156.f81d1b38@mail.tritonwest.net> Getting the calls parked is the right way, but there is an easier way to do it. The two legs that you want to remain in the call after the conference bridge, you can just uuid_transfer them both to park as an inline dialplan right in the transfer, then you're free to do the uuid_bridge. The variable "park_after_bridge" is used in the core by switch_ivr_bridge() funciton. It really only applies to a call that's been bridged via that function, which would not apply in all cases. For example if you called conference form the dialplan, or some other dp app that doesn't use "switch_ivr_bridge". It also checks the variable on terminating the bridge. This allows you to set the variable after the fact, while the channel is bridged. However, since the call may of not started out bridged, the park_after_bridge will not have any meaning. So just to be safe, and not have to track every call via ESL, to know how it got to the state, I would do a "uuid_transfer park inline" to each channel you want to remove from the conference, and then you can do the "uuid_bridge". --Dave _____ From: Tihomir Culjaga [mailto:tculjaga at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 27 May 2013 08:21:02 -0700 Subject: Re: [Freeswitch-users] bridging 2 call legs after a conference ya, i have park_after_bridge=true on the incoming call leg :=) ... thats why it goes on park after leaving the conference... never thought park_after_bridge affects conference as well ... :) thanks anyway. T. On Mon, May 27, 2013 at 4:13 PM, Tihomir Culjaga wrote: hello folks, im controlling FS via ESL and what im trying to do is to make a 3-way conference and this works fine. What i have an issue with is when one conference member hangs up, i want to return back to a normal bridge between the 2 remaining members ... so what im doing is: On Incoming call from 1013 to FS answer the call: ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: answer"]) play an announcement to A leg: CMD uuid_broadcast "bgapi uuid_broadcast 00349e91-117e-4dcc-b2db-45d3a54c5fee 'greeting1.wav'" originate a new call and parks it: CMD originate "bgapi originate {origination_uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234,ignore_early_media=false,originate_timeout=15}user/1002 &park" bridge these 2 legs into a call: CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" originate a 2nd new call into a conference: CMD originate "bgapi originate {origination_uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f,ignore_early_media=false,originate_timeout=0}user/1014 &conference(supervizor36 at silent+flags{mute}) &park" join the 1st new call (user/1002) CMD_MULTI conference ["sendmsg fc3e44c5-63af-45bb-8a8e-8a35bac52234","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36 at silent"] join incoming call into the same conference CMD_MULTI conference ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36 at silent"] unmute uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 unmute 20" Incoming call 00349e91-117e-4dcc-b2db-45d3a54c5fee must not hear uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 relate 21 20 nohear" so far its great! now, call uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f (the one originated into the conference) hangs up, 2 remaining calls (incoming call and 1st originated calls) can still hear each others... but i want to get rid of a conference since there are just 2 members hence no sense keeping the conference up... so i do: bridge the two remaining conference members together CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" and i get no audio!! looks like FS setting both channels on park ?? any idea ? * 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1367 Session 30 (sofia/internal/sip:1014 at 192.168.254.112:5060) Ended * 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:1014 at 192.168.254.112:5060 [CS_DESTROY] * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1014 at 192.168.254.112:5060) Callstate Change HANGUP -> DOWN * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1014 at 192.168.254.112:5060) Running State Change CS_DESTROY * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY * 2013-05-27 15:34:13.724037 [DEBUG] mod_sofia.c:363 sofia/internal/sip:1014 at 192.168.254.112:5060 SOFIA DESTROY * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1014 at 192.168.254.112:5060 Standard DESTROY * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY going to sleep * 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1521 (sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_SOFT_EXECUTE -> CS_HIBERNATE * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1523 (sofia/external/1013 at 192.168.254.93) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE * 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/internal/sip:1002 at 192.168.254.116:5060 Restore previous codec PCMA:8. * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/external/1013 at 192.168.254.93 Restore previous codec PCMA:8. * 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1602 Write Lock ON * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1002 at 192.168.254.116:5060] * 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1605 Write Lock OFF * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:1366 (sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_HIBERNATE -> CS_RESET * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE going to sleep * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change CS_RESET * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET * 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1002 at 192.168.254.116:5060 SOFIA RESET * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:721 sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM RESET * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:728 (sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_RESET -> CS_SOFT_EXECUTE * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET going to sleep * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State Change CS_SOFT_EXECUTE * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE * 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:572 SOFIA SOFT_EXECUTE * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:746 sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM SOFT_EXECUTE * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/1013 at 192.168.254.93] * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr.c:2422 (sofia/external/1013 at 192.168.254.93) State Change CS_HIBERNATE -> CS_PARK * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013 at 192.168.254.93 [BREAK] * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:420 (sofia/external/1013 at 192.168.254.93) State EXCHANGE_MEDIA going to sleep * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1013 at 192.168.254.93) Running State Change CS_PARK * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:426 (sofia/external/1013 at 192.168.254.93) State PARK * 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:247 sofia/external/1013 at 192.168.254.93 Standard PARK * * * * freeswitch at internal> 2013-05-27 15:34:18.457165 [WARNING] sofia_reg.c:1400 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1014 at 192.168.254.93]from ip 192.168.254.112 * * freeswitch at internal> * freeswitch at internal> * freeswitch at internal> show calls * uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch * 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49,1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168.254.112,38515494471,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,cc01,Outbound Call,1002,,,,,,,,,,,,,,,,,,, * fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05,1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,1002 at 192.168.254.93,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,cc01,1013,1013,,,,,,,,,,,,,,,,,,, * * 2 total. * * freeswitch at internal> show channels * uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num * 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49,1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013,192.168.254.112,38515494471,conference,supervizor36 at silent,XML,default,PCMA,8000,64000,PCMA,8000,64000,,cc01,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,Outbound Call,1002 * fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05,1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,conference,supervizor36 at silent,,default,PCMA,8000,64000,PCMA,8000,64000,,cc01,1002 at 192.168.254.93,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,1013,1013 * * 2 total. * * freeswitch at internal> * freeswitch at internal> * freeswitch at internal> * freeswitch at internal> 2013-05-27 15:34:57.732290 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/3114c1e1/attachment-0001.html From bigx333 at gmail.com Mon May 27 20:35:15 2013 From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado) Date: Mon, 27 May 2013 18:35:15 +0200 Subject: [Freeswitch-users] Freeswitch just won't transcode the call. In-Reply-To: References: Message-ID: The G729 that comes with FreeSWITCH is pass through only, if you want transcoding you need to buy a license. On 27 May 2013 3:55 PM, "Daniel Ivanov" wrote: > I'm having media problems with a FS here. UAs are sending mixed > capabilites all the time and i wanna gracefully degradate on them. > > This is what i have in vars.xml: > > > > This is in external.xml: > > > > > > > I need to make sure that when a UA dials in, the A-leg is negotiated > between g729 and gsm(and codec2) and the call is then transcoded to the > B-leg if it's not G729 instead of just resending the SDP directly to the > gateways. > I've tried disabling and enabling proxy-media, late-negotiation and > disable-transcoding along with passing the strict codec_string and > absolute_codec_string. > > Point me on what i'm failing to apprehend here. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/be56b0f9/attachment.html From grcamauer at gmail.com Mon May 27 20:43:41 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Mon, 27 May 2013 13:43:41 -0300 Subject: [Freeswitch-users] Freeswitch just won't transcode the call. In-Reply-To: References: Message-ID: If you are going to be running many simultaneous calls, you might want to look into a Digium TCE400B or Sangoma D100 card. You get the G729 licences and offload the CPU at the same time. Software licenses are better if you need just a few. Guillermo On Mon, May 27, 2013 at 1:35 PM, Nelson Luiz Ferraz de Camargo Penteado < bigx333 at gmail.com> wrote: > The G729 that comes with FreeSWITCH is pass through only, if you want > transcoding you need to buy a license. > On 27 May 2013 3:55 PM, "Daniel Ivanov" wrote: > >> I'm having media problems with a FS here. UAs are sending mixed >> capabilites all the time and i wanna gracefully degradate on them. >> >> This is what i have in vars.xml: >> >> >> >> This is in external.xml: >> >> >> >> >> >> >> I need to make sure that when a UA dials in, the A-leg is negotiated >> between g729 and gsm(and codec2) and the call is then transcoded to the >> B-leg if it's not G729 instead of just resending the SDP directly to the >> gateways. >> I've tried disabling and enabling proxy-media, late-negotiation and >> disable-transcoding along with passing the strict codec_string and >> absolute_codec_string. >> >> Point me on what i'm failing to apprehend here. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/fd507af7/attachment.html From xiaofengcanyuexp at 163.com Mon May 27 20:45:04 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Tue, 28 May 2013 00:45:04 +0800 Subject: [Freeswitch-users] sip_ph_X- header doesn't work for freetdm-sofia call Message-ID: <201305280045030372439@163.com> Dear support I got a problem to pass through SS7 info in SIP messages. Here is the code change and test call flow: -------------------- Code change and log: -------------------- 1. At sangoma module:intercept ACM in ftmod_sangoma_ss7_*.c, parse the BCI and add it to the ftdmchan->call_data: sngss7_add_var(sngss7_info, "ss7_acm_bci_hex", val); 2. At freetdm module mod_freetdm.c, get the variable and set it to channel. Of which, the "prefix" is 'sip_ph_X-': var_value = ftdm_sigmsg_get_var(sigmsg, "ss7_acm_bci_hex"); if (!ftdm_strlen_zero(var_value)) { snprintf(variable_name, "%sTEST-BCI-HEX", prefix); switch_channel_set_variable_printf(channel, variable_name, "%s", var_value); } >From the test call log: 2013-05-27 07:03:47.747361 [INFO] ftmod_sangoma_ss7_handle.c:362 [s1c1][1:1] [CIC:1]Rx ACM 2013-05-27 07:03:47.747361 [DEBUG] ftmod_sangoma_ss7_support.c:2920 [s1c1][1:1] ACM BCI parameter Hex: 0x1636 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5513 Get ACM BCI parameter from sangoma 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5530 sip_ph_X-TEST-BCI-HEX:1636 --------------------- My purpose: --------------------- What I want to do is the encapsulate the sip_ph_X-TEST-BCI-HEX in the 183 SIP message. Like handling IAM in ftdm_channel_from_event(), once it sets the the variable like below, the "X-FreeTDM-SpanNumber" will automatically attached in INVITE message. switch_channel_set_variable_printf(channel, "sip_h_X-FreeTDM-SpanNumber", "%d", spanid); --------------------- Problems: --------------------- While, what's strange are: 1. I add one line in dialplan/default.xml to log the variable with: The freeswitch.log show ${sip_ph_X-TRUSTID-BCI-HEX} is empty 2. The 183 message does not attach the 'X-TRUSTID-BCI-HEX'. Can someone help to figure out what's wrong with it? Thanks ------------------- 2013-05-28 From lists at telefaks.de Mon May 27 20:53:34 2013 From: lists at telefaks.de (Peter Steinbach) Date: Mon, 27 May 2013 18:53:34 +0200 Subject: [Freeswitch-users] WebRTC In-Reply-To: References: Message-ID: <51A38F8E.5050409@telefaks.de> I am also very curious to see it working. It seems that WebRTC could drive the next generation of VoIP phones like e.g. these: http://www.hkaie.com/list/?114_1.html We'll just need a tablet, a handset, freeswitch and a webserver to load the phone application from. Best regards Peter On 05/24/13 20:51, mbo wrote: > I've seen there have been a couple of discussions about WebRTC support earlier this year. Are there any news on that? > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From drk at drkngs.net Mon May 27 20:58:58 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Mon, 27 May 2013 09:58:58 -0700 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: Message-ID: <20130527165858.28916fed@mail.tritonwest.net> This is a major can of worms. I have an comericial carrier softswitch based on FS, and what I did to be able to imlment billing the right legs to the right account was the following: Have a flag on your gateways so you can mark them as "pstn-uplink", or whatever you want to call it. This way you know that it's the "other" leg's endpoint is the entity you want to bill. Next, in your directory, make sure you set a variable for the user that has it's billing identity for example "myapp_user" or "myapp_account". If you use account then you can associate more then one user to the same account, if that's the way you're billing works. Now, if you make sure all calls to users, use the user/username at domain form of the dial string, you will also have the variable set on calls terminating to that user. Now you can bill, on normal calls both incoming calls (charge for DID usage) and outbound calls. When you get any inbound call from a gateway that's marked as "PSTN" then set a variable on that leg, when it calls a user, that indicates it's the PAID inbound leg for that user, before bridging to the "user" channel. To implment forwarding, in your processing of a call to a user, before bridging to the "user" channel, see if the users is forwarding, if so all you have to do, in your dialplan, or external code handling the dialplan is: "set_user to the user they called", then "transfer to: ${Forwarding_number} XML ${user_context}". Because you did the "set_user" first, the call will go back to CS_ROUTING, but look like it's from the user that did the forwarding. This way, if the call is from a gateway that isn't in the right context to dial the forwarding number, it won't matter, and the B-LEG will go through your dialplan logic, like the call is made form the user, and you can set a variable on the B_LEG in the dial string to tell it who the call is on behalf of, if it goes out a gateway that is marked as "PSTN". This way the same logic of how you tag the PSTN leg as being called from a user, is the same for a call made from the user, or forwared by the user. Since any users in the middle that are forwarding to internal numbers, it doesn't matter how many forwarding hops you go through, the original A-LEG if PSTN will get billed to the user it called, and the final B-LEG will get biled to the user that did the forwarding, that caused the call to leave your switch. --Dave _____ From: Jon Sch?pzinsky [mailto:jos at firstcom.dk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 27 May 2013 09:01:49 -0700 Subject: [Freeswitch-users] Splitting CDRs on call forward Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Sch?pzinsky _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130527/0667fd74/attachment-0001.html From anthony.minessale at gmail.com Mon May 27 21:07:35 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 May 2013 12:07:35 -0500 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: Message-ID: Funny, forkcdr.... Guess who made that ;) That was my early solution to billing problems that I later solved by writing FS! Yes ken is right, log the B legs too and you will see both ends of the call. Also look at xml cdrs and custom cdr bindings. On May 27, 2013 11:31 AM, "Ken Rice" wrote: > if you are lusing xml cdr, set the logging to both not just a or b legs, > this will generate cdrs for all legs individually > > Ken > Sent from my iPad > > On May 27, 2013, at 11:01, Jon Sch?pzinsky wrote: > > > Hi List, > > > > I am implementing call forwarding on a multi tenant system, and therefore > > need to split CDR's when the call forward happens, so that if the > > receiving user also has his account call forwarded, he pays for his part > > of the call. > > > > A calls B > > B forwards to C > > C forwards to an external mobile phone. > > > > B has a free call from B to C, but C needs to pay for the forwarding to > > the mobile phone. Therefore i need a separate CDR for the C to Mobile > > phone call. > > > > Another example would be this > > > > A works in Company A, and B works in Company B > > They are both users on our system, and therefore is on the same > freeswitch. > > > > A calls B > > B Forwards to an external mobile phone. > > > > Here A needs to pay for the call from A to B, and B needs to pay for the > > call being forwarded to his mobile phone. > > > > Do anybody have an idea as to how to implement this in freeswitch. Back > in > > my Asterisk days, this would be done by the ForkCDR command. > > > > > > Venlig hilsen/kind regards > > > > Jon Leren Sch?pzinsky > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/c2ce459b/attachment.html From sertys at gmail.com Mon May 27 21:10:57 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Mon, 27 May 2013 19:10:57 +0200 Subject: [Freeswitch-users] Freeswitch just won't transcode the call. In-Reply-To: References: Message-ID: Thank you for reminding me. I am gonna use g729 less and less from now on, but i have a few licences on that machine for transcoding it, so this was not the problem. As it happens in many cases , you have to find the solution yourself just 3 minutes after writing a long thread to the maillist. What i have been missing was the inbound-codec-negotiation that i have been setting to scrooge(just because it sounds funny). And this has been sending the same SDP as from the A-leg. By the time i reverted to generous it started working better. On Mon, May 27, 2013 at 6:43 PM, Guillermo Ruiz Camauer wrote: > If you are going to be running many simultaneous calls, you might want to > look into a Digium TCE400B or Sangoma D100 card. You get the G729 > licences and offload the CPU at the same time. Software licenses are > better if you need just a few. > > Guillermo > > > On Mon, May 27, 2013 at 1:35 PM, Nelson Luiz Ferraz de Camargo Penteado < > bigx333 at gmail.com> wrote: > >> The G729 that comes with FreeSWITCH is pass through only, if you want >> transcoding you need to buy a license. >> On 27 May 2013 3:55 PM, "Daniel Ivanov" wrote: >> >>> I'm having media problems with a FS here. UAs are sending mixed >>> capabilites all the time and i wanna gracefully degradate on them. >>> >>> This is what i have in vars.xml: >>> >>> >>> >>> This is in external.xml: >>> >>> >>> >>> >>> >>> >>> I need to make sure that when a UA dials in, the A-leg is negotiated >>> between g729 and gsm(and codec2) and the call is then transcoded to the >>> B-leg if it's not G729 instead of just resending the SDP directly to the >>> gateways. >>> I've tried disabling and enabling proxy-media, late-negotiation and >>> disable-transcoding along with passing the strict codec_string and >>> absolute_codec_string. >>> >>> Point me on what i'm failing to apprehend here. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/1745304e/attachment.html From cal.leeming at simplicitymedialtd.co.uk Mon May 27 21:16:17 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Mon, 27 May 2013 18:16:17 +0100 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: Message-ID: You have to identify the correct headers dependent on the call context (i.e. which leg did the bridge/forward, the originatee/originator etc). Personally I've found that attaching custom headers onto each leg is the best way to keep track things - but like Dave said, this really is a can of worms. It took me approx 6 months of various code iterations to eventually figure out a way to do this that was suitable for our use case. Several people have said to me "Cal, it's harder than it looks"... tbh it isn't, it's just that the variables and API aren't exposed in a typically sane fashion, and there is no definitive guide on the patterns to look for or best practices. In short, it isn't complex, but it might make you bang your head off a wall out of sheer frustration. You need to ensure you compensate for both blind and attended transfers, and consider what happens if a transfer goes between two domains.. that's when things get really sticky :) Cal On Mon, May 27, 2013 at 5:01 PM, Jon Sch?pzinsky wrote: > Hi List, > > I am implementing call forwarding on a multi tenant system, and therefore > need to split CDR's when the call forward happens, so that if the > receiving user also has his account call forwarded, he pays for his part > of the call. > > A calls B > B forwards to C > C forwards to an external mobile phone. > > B has a free call from B to C, but C needs to pay for the forwarding to > the mobile phone. Therefore i need a separate CDR for the C to Mobile > phone call. > > Another example would be this > > A works in Company A, and B works in Company B > They are both users on our system, and therefore is on the same freeswitch. > > A calls B > B Forwards to an external mobile phone. > > Here A needs to pay for the call from A to B, and B needs to pay for the > call being forwarded to his mobile phone. > > Do anybody have an idea as to how to implement this in freeswitch. Back in > my Asterisk days, this would be done by the ForkCDR command. > > > Venlig hilsen/kind regards > > Jon Leren Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/3f4df4b1/attachment-0001.html From krice at freeswitch.org Mon May 27 21:20:50 2013 From: krice at freeswitch.org (Ken Rice) Date: Mon, 27 May 2013 12:20:50 -0500 Subject: [Freeswitch-users] OpenVZ tuning tips In-Reply-To: Message-ID: The reason for the move off of centos is widely reported... There were issues with performance we couldn?t track down that only happened on centos6 On 5/27/13 5:51 AM, "Tamas Jalsovszky" wrote: > Hello Ken, > > As I've checked, there is new proxmox version based on Wheezy. As I've found, > Proxmox kernel is based on RHEL6x, so probably it needs intensive testing too. > What were the reasons you decided to move off from CentOS? Were you able to > reproduce problems? > > Br, > ? Tamas > > > On Sun, May 26, 2013 at 8:12 PM, Ken Rice wrote: >> I actually use Proxmox VE which provides a nice web ui for managing both >> OpenVZ and KVM instances. Proxmox VE is built on top of Debian Squeeze (not >> sure if they have started moving their dev to wheezy at this point) >> >> it works well. many services on the freeswitch.org >> domain are ran this way also. >> >> one of the things i use proxmox to accomplish is host managament. i'm running >> the entire box dedicated to 1 FS instance, but still use openvz so i can move >> an instance to a backup box incase of issues >> >> >> >> Ken >> Sent from my iPad >> >> On May 26, 2013, at 12:09, Tamas Jalsovszky wrote: >> >>> Hello Ken, >>> >>> Do you use any kind of virtualizaztion/separation on Debian? We would like >>> to keep lighweight 'virtualization' with openvz or maybe lxc (or any other >>> idea?). Any experience with that on Debian? >>> Is Debian better suited for FS than Ubuntu LTS (e.g. 12.04)? >>> It seems, we have to find out where to go from latest centos5+ovz+ubuntu >>> 10.04 in VE... >>> >>> T. >>> >>> >>> On Sat, May 25, 2013 at 4:42 PM, Ken Rice wrote: >>>> I actually dropped using Centos6 and moved to debian to get both timerfd >>>> and to get performance at the same time. And things started working much >>>> nicer... Not sure if they even fixed the performance issues on centos tho >>>> >>>> >>>> >>>> On 5/25/13 4:52 AM, "Yuriy Nasida" >>>> > wrote: >>>> >>>>> Tamas, >>>>> >>>>> I think you can find info about performance problems with CentOS 6 on >>>>> jira. >>>>> For example. http://jira.freeswitch.org/browse/FS-4291 >>>>> >>>>> We also wanted to use timerfd (without virtualization) and ?made a moving >>>>> to latest cenos 6.(2,3) + FS 1.2.8 . It was big mistake. FS got frozen >>>>> sometimes. As a result we had to move on centos 5.9 again. >>>>> >>>>> I would like to join issue. how can I be sure that timerfd is used? >>>>> >>>>> Regards, >>>>> Yuriy >>>>> >>>>> >>>>> Date: Sat, 25 May 2013 07:52:28 +0200 >>>>> From: jalsot at gmail.com >>>>> To: freeswitch-users at lists.freeswitch.org >>>>> >>>>> Subject: Re: [Freeswitch-users] OpenVZ tuning tips >>>>> >>>>> We've tried as root user but as I remember it was not able to set up the >>>>> priority but will recheck to be sure. Another interesting thing is that >>>>> when I tried to set a running process scheduler with chrt, got operation >>>>> not permitted (as root of course), so I guess, something has to be tuned >>>>> in the VE or on the host. We will try on bare metal centos6+ovz on the >>>>> host. >>>>> We try the latest centos6 with latest openvz kernel (due to security >>>>> requirements we run on latest stable kernel and OS versions). >>>>> Could you give some info about those horrible performance problems to let >>>>> us check whether it still applies or not? (we've found only mysql create >>>>> table performance degradation due to ext4 - where the solution could be >>>>> barrier=0 yet, no other problems). >>>>> Actually, how can I be sure that timerfd is used? strace? I'm nearly sure >>>>> that timerfd works fine in FS. >>>>> >>>>> Yep, it would be much simpler without virtualization, and much harder from >>>>> another perspective. Probably lxc, kvm and xen aren't much better >>>>> regarding realtime stuff... >>>>> >>>>> Regards, >>>>> ???Tamas >>>>> >>>>> >>>>> On Fri, May 24, 2013 at 11:28 PM, Anthony Minessale >>>>> > wrote: >>>>>> It will only work running as root I believe because it needs high privs >>>>>> to do realtime. >>>>>> If you do use centOS make sure its the latest rev of cent6, we have some >>>>>> horrible performance problems on the earlier revs and I don't know if >>>>>> they were resolved. >>>>>> Just make sure your kernels are as new as you can get them on the real >>>>>> host to avoid years of kernel performance bugs and that it has support >>>>>> for timerfd otherwise your VE would compile with timerfd support but not >>>>>> have actual access to the real syscalls for it in the host. >>>>>> >>>>>> That's why we try not to recommend virtual stuff in general as it takes >>>>>> some very careful setups and its hard to support from our standpoint when >>>>>> people run into issues. >>>>>> >>>>>> ? >>>>>> >>>>>> >>>>>> >>>>>> On Fri, May 24, 2013 at 12:30 AM, Tamas Jalsovszky >>>>> > wrote: >>>>>>> Hello, >>>>>>> >>>>>>> Thank you for tips, we are testing centos/openvz 6 with 2.6.32 kernel on >>>>>>> host and Ubuntu 10.04 LTS in VE. >>>>>>> Do you know maybe how to allow realtime priority in the VE for FS? >>>>>>> Running FS with -rp does not set the scheduler. >>>>>>> strace says, sched_setscheduler operation permitted, so SCHED_FF is not >>>>>>> set. Tried to run as root and/or use ulimit -r option, but cannot run FS >>>>>>> with tuned priorities.We guess, some thing missing in the host/VE >>>>>>> configuraton. >>>>>>> Any idea? >>>>>>> >>>>>>> Br, >>>>>>> ??Tamas >>>>>>> >>>>>>> >>>>>>> On Thu, May 23, 2013 at 12:47 AM, jay binks >>>>>> > wrote: Im using 2.6.32 on all my boxes ... One thing that has me thinking, ?are there any tweaks to get MSI-X working best it can ? ( with proxmox ) there seems to be a strong bias towards one CPU for all interrupts. I could be wrong, but its something I think ive seen, and didnt see any clear suggestions on. Jay On 23 May 2013 01:12, Anthony Minessale > wrote: 2.6.25 or newer to get timerfd support. On Wed, May 22, 2013 at 2:56 AM, Zenny > wrote: On 5/22/13, Anthony Minessale > wrote: > You should consider centos6 or debian stable. ?Make sure the host kernel is > very new to get maximum results. Tony, do you mean "very new kernel" means 3.2.xx kernel? Openvz host kernel is still at 2.6.32 so bleeding edge kernel is not possible. And that is what CentOS6 offers, too. However, I installed FS as openvz guest, it works fine for outgoing, but not DNAT works for incoming connections even after throroughly following http://wiki.freeswitch.org/wiki/NAT_Traversal#FreeSWITCH_behind_NAT. Just my two cents. > > > On Tue, May 21, 2013 at 2:53 PM, Tamas Jalsovszky > wrote: > >> Hello, >> >> Do you have any recommendations regarding how to set up correctly (for >> production) CentOS5 openvz and FS 1.2.stable? Is there any trick to >> tuneup >> the system to be rock solid? >> Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS >> 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that >> something is not well configured around timers, however mod_posix_timer >> did >> not help anything (running FS with -rp). We use our own bare metal and >> can >> reproduce those delatas eirher when only one VE is on the HW. >> Maybe time to check out centos6 with openvz? >> >> Any idea, recommendation, experience can be very helpful. >> >> Regards, >> ???Jalsot >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net ?#freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/855f9dec/attachment.html From a.mykhalkiv at kwebbl.com Mon May 27 17:28:03 2013 From: a.mykhalkiv at kwebbl.com (Anatolii) Date: Mon, 27 May 2013 06:28:03 -0700 (PDT) Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) Message-ID: <1369661283368-7591135.post@n2.nabble.com> Hello all! Freeswitch version - 1.2.8, installed in Centos. Freeswitch always goes for configuration to our server (on which CakePHP is situated) I have problem with callcenter module in freeswitch. In mod_xml_curl add two bindings: dialplan and configuration. < binding name="configuration"> < param name="use-dynamic-url" value="yes"/> < param name="gateway-url" value="http://example.comt/freeswitch/configuration/${our_queue_name}${company_id}" bindings="configuration"/> < /binding> So, when I restart (start) freeswitch or reload mod_callcenter - FS request default configuration for callcenter and it works great (FS add queues, agents and tiers). After that in dialplan for specific number I set application callcenter with specific name and set 2 global variables, *our_queue_name* and *company_id*, which send to me with a name of queue and company id. -- this works fine When I receive from freeswitch request - I configure xml from db data and send FS callcenter configuration. -- works fine. *FS get correct xml, but add only queue settings and absolutely ignoring adding agents and tiers.* -- This is my problem for now. I try to solve this situation by using lua scripts - no results. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-callcenter-configuration-don-t-register-agents-and-tiers-tp7591135.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/052e5c6f/attachment-0001.html From jos at firstcom.dk Mon May 27 22:29:10 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Mon, 27 May 2013 20:29:10 +0200 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: <20130527165858.28916fed@mail.tritonwest.net> References: , <20130527165858.28916fed@mail.tritonwest.net> Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B23E34FE518@helle.dk.hq.firstcom.dk> Hello Dave, The main problem isnt really the first scenario, as only the last leg needs to be billed. The biggest issue is with the second scenario, since both calls need to be billed, as theres two billing entities involved. I am also not using the FreeSWITCH directory, as the authentication and registration are handled by an upstream OpenSIPS instead. But thank you for the very thorough description. Its definitely inspiring working towards a solution. Right now im leaning towards using hooks, and taking care of the billing in lua, instead of using the internal CDR system. That way i can get it exactly as i want it :) /Jon ________________________________________ Fra: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] På vegne af Dave R. Kompel [drk at drkngs.net] Sendt: 27. maj 2013 18:58 Til: FreeSWITCH Users Help Emne: Re: [Freeswitch-users] Splitting CDRs on call forward This is a major can of worms. I have an comericial carrier softswitch based on FS, and what I did to be able to imlment billing the right legs to the right account was the following: Have a flag on your gateways so you can mark them as "pstn-uplink", or whatever you want to call it. This way you know that it's the "other" leg's endpoint is the entity you want to bill. Next, in your directory, make sure you set a variable for the user that has it's billing identity for example "myapp_user" or "myapp_account". If you use account then you can associate more then one user to the same account, if that's the way you're billing works. Now, if you make sure all calls to users, use the user/username at domain form of the dial string, you will also have the variable set on calls terminating to that user. Now you can bill, on normal calls both incoming calls (charge for DID usage) and outbound calls. When you get any inbound call from a gateway that's marked as "PSTN" then set a variable on that leg, when it calls a user, that indicates it's the PAID inbound leg for that user, before bridging to the "user" channel. To implment forwarding, in your processing of a call to a user, before bridging to the "user" channel, see if the users is forwarding, if so all you have to do, in your dialplan, or external code handling the dialplan is: "set_user to the user they called", then "transfer to: ${Forwarding_number} XML ${user_context}". Because you did the "set_user" first, the call will go back to CS_ROUTING, but look like it's from the user that did the forwarding. This way, if the call is from a gateway that isn't in the right context to dial the forwarding number, it won't matter, and the B-LEG will go through your dialplan logic, like the call is made form the user, and you can set a variable on the B_LEG in the dial string to tell it who the call is on behalf of, if it goes out a gateway that is marked as "PSTN". This way the same logic of how you tag the PSTN leg as being called from a user, is the same for a call made from the user, or forwared by the user. Since any users in the middle that are forwarding to internal numbers, it doesn't matter how many forwarding hops you go through, the original A-LEG if PSTN will get billed to the user it called, and the final B-LEG will get biled to the user that did the forwarding, that caused the call to leave your switch. --Dave ________________________________ From: Jon Sch?pzinsky [mailto:jos at firstcom.dk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 27 May 2013 09:01:49 -0700 Subject: [Freeswitch-users] Splitting CDRs on call forward Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Sch?pzinsky _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- This message has been scanned for viruses and dangerous content, and is believed to be clean. From jos at firstcom.dk Mon May 27 22:33:35 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Mon, 27 May 2013 20:33:35 +0200 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: , Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B23E34FE519@helle.dk.hq.firstcom.dk> My primary problem is that the legs inbetween the A leg and the C leg isnt actually bridged calls, but all done in dialplan, which makes the second scenario especially difficult. But I am going to try and code my way out if it, using api hooks and lua. That way i can get exactly what i need for the billing. /Jon ________________________________________ Fra: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] På vegne af Anthony Minessale [anthony.minessale at gmail.com] Sendt: 27. maj 2013 19:07 Til: FreeSWITCH Users Help Emne: Re: [Freeswitch-users] Splitting CDRs on call forward Funny, forkcdr.... Guess who made that ;) That was my early solution to billing problems that I later solved by writing FS! Yes ken is right, log the B legs too and you will see both ends of the call. Also look at xml cdrs and custom cdr bindings. On May 27, 2013 11:31 AM, "Ken Rice" > wrote: if you are lusing xml cdr, set the logging to both not just a or b legs, this will generate cdrs for all legs individually Ken Sent from my iPad On May 27, 2013, at 11:01, Jon Sch?pzinsky > wrote: > Hi List, > > I am implementing call forwarding on a multi tenant system, and therefore > need to split CDR's when the call forward happens, so that if the > receiving user also has his account call forwarded, he pays for his part > of the call. > > A calls B > B forwards to C > C forwards to an external mobile phone. > > B has a free call from B to C, but C needs to pay for the forwarding to > the mobile phone. Therefore i need a separate CDR for the C to Mobile > phone call. > > Another example would be this > > A works in Company A, and B works in Company B > They are both users on our system, and therefore is on the same freeswitch. > > A calls B > B Forwards to an external mobile phone. > > Here A needs to pay for the call from A to B, and B needs to pay for the > call being forwarded to his mobile phone. > > Do anybody have an idea as to how to implement this in freeswitch. Back in > my Asterisk days, this would be done by the ForkCDR command. > > > Venlig hilsen/kind regards > > Jon Leren Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- This message has been scanned for viruses and dangerous content, and is believed to be clean. From jos at firstcom.dk Mon May 27 22:35:40 2013 From: jos at firstcom.dk (=?iso-8859-1?Q?Jon_Sch=F8pzinsky?=) Date: Mon, 27 May 2013 20:35:40 +0200 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: , Message-ID: <111509FCE9EBF74BAB2CB38AAC521D9B23E34FE51A@helle.dk.hq.firstcom.dk> A certain amount of banging heads against walls has already happened, because as you said, its actually pretty basic, but theres so many scenarios to consider that it gets advanced, fast. But at least its fun with FreeSWITCH :) /Jon ________________________________________ Fra: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] På vegne af Cal Leeming [Simplicity Media Ltd] [cal.leeming at simplicitymedialtd.co.uk] Sendt: 27. maj 2013 19:16 Til: FreeSWITCH Users Help Emne: Re: [Freeswitch-users] Splitting CDRs on call forward You have to identify the correct headers dependent on the call context (i.e. which leg did the bridge/forward, the originatee/originator etc). Personally I've found that attaching custom headers onto each leg is the best way to keep track things - but like Dave said, this really is a can of worms. It took me approx 6 months of various code iterations to eventually figure out a way to do this that was suitable for our use case. Several people have said to me "Cal, it's harder than it looks"... tbh it isn't, it's just that the variables and API aren't exposed in a typically sane fashion, and there is no definitive guide on the patterns to look for or best practices. In short, it isn't complex, but it might make you bang your head off a wall out of sheer frustration. You need to ensure you compensate for both blind and attended transfers, and consider what happens if a transfer goes between two domains.. that's when things get really sticky :) Cal On Mon, May 27, 2013 at 5:01 PM, Jon Sch?pzinsky > wrote: Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Sch?pzinsky _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- This message has been scanned for viruses and dangerous content, and is believed to be clean. From steveayre at gmail.com Mon May 27 22:53:10 2013 From: steveayre at gmail.com (Steven Ayre) Date: Mon, 27 May 2013 19:53:10 +0100 Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION In-Reply-To: References: <1369468901841-7591099.post@n2.nabble.com> Message-ID: > > recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: > ------------------------------------------------------------ > ------------ > SIP/2.0 606 Not Acceptable Yes. The 606 packet is generated by the goip and sent by the goip to freeswitch. We/FreeSWITCH cannot tell you why the goip sent 606, you would need to check its logs/records or with the provider/manufacturer to see why. The only thing I would say is a codec mismatch is normally 488 not 606, so chances are it's something else... but that is assuming that they are following the standards, which is not necessarily the case. One thing I would try is setting the verbose_sdp variable: See http://wiki.freeswitch.org/wiki/Variable_verbose_sdp for -Steve On 26 May 2013 12:20, eoaddai wrote: > So, is the call hitting the goip at all? or, freeswitch is unable to > process it to the goip? > > > On 26 May 2013 10:32, Daniel Ivanov [via freeswitch-users] <[hidden email] > > wrote: > >> Maybe you're not sending them the right codecs or trying to run a feature >> they don't have. Revisit your vars.xml and sip_profiles .xml. . Ultimately >> contact the provider to ask them what youre doing wrong in your sdps. >> On May 26, 2013 11:26 AM, "eoaddai" <[hidden email]> >> wrote: >> >>> Hi, my first time of posting stuff here. I really need help. >>> I keep on not getting my calls go through freeswitch to my goip gateway. >>> The >>> following is the freeswitch log with siptrace turned on i get. Please >>> help >>> me: >>> >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing >>> global variables >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [origination_caller_id_number]=[0264370536] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_from]=[0264370536] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_to]=[0267577771] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_app]=[true] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [originate_timeout]=[60] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>> [ignore_early_media]=[true] >>> 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel >>> sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] >>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 >>> (sofia/external/0267577771) State Change CS_NEW -> CS_INIT >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0267577771) Running State Change CS_INIT >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/external/0267577771) State INIT >>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 >>> sofia/external/0267577771 >>> SOFIA INIT >>> 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: >>> v=0 >>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>> s=FreeSWITCH >>> c=IN IP4 10.10.50.1 >>> t=0 0 >>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> a=sendrecv >>> >>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 >>> (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/external/0267577771) State INIT going to sleep >>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: >>> >>> ------------------------------------------------------------------------ >>> INVITE [hidden email]SIP/2.0 >>> >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>> Max-Forwards: 70 >>> From: "" <[hidden email] >>> >;tag=DXjN0tK6mcc5S >>> To: <[hidden email] >>> > >>> >>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 INVITE >>> Contact: >>> User-Agent: >>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, hold, conference, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 201 >>> X-FS-Support: update_display,send_info >>> Remote-Party-ID: >>> <[hidden email] >>> >;party=calling;screen=yes;privacy=off >>> >>> >>> v=0 >>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>> s=FreeSWITCH >>> c=IN IP4 10.10.50.1 >>> t=0 0 >>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> >>> ------------------------------------------------------------------------ >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0267577771) Running State Change CS_ROUTING >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/external/0267577771) State ROUTING >>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 >>> sofia/external/0267577771 >>> SOFIA ROUTING >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 >>> (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/external/0267577771) State ROUTING going to sleep >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>> (sofia/external/0267577771) State CONSUME_MEDIA >>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>> (sofia/external/0267577771) State CONSUME_MEDIA going to sleep >>> 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel >>> sofia/external/0267577771 entering state [calling][0] >>> recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 606 Not Acceptable >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>> From: "" <[hidden email] >>> >;tag=DXjN0tK6mcc5S >>> To: <[hidden email] >>> >;tag=1662509363 >>> >>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 INVITE >>> User-Agent: dble >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: >>> >>> ------------------------------------------------------------------------ >>> ACK [hidden email]SIP/2.0 >>> >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>> Max-Forwards: 70 >>> From: "" <[hidden email] >>> >;tag=DXjN0tK6mcc5S >>> To: <[hidden email] >>> >;tag=1662509363 >>> >>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 ACK >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel >>> sofia/external/0267577771 entering state [terminated][606] >>> 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup >>> sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 Originate >>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal >>> sofia/external/0267577771 [KILL] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0267577771) Running State Change CS_HANGUP >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/external/0267577771) State HANGUP >>> 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel >>> sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 >>> sofia/external/0267577771 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/external/0267577771) State HANGUP going to sleep >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 >>> (sofia/external/0267577771) Callstate Change DOWN -> HANGUP >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 >>> (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0267577771) Running State Change CS_REPORTING >>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 >>> (sofia/external/0267577771) State REPORTING >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 >>> sofia/external/0267577771 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 >>> (sofia/external/0267577771) State REPORTING going to sleep >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0267577771 [BREAK] >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 >>> (sofia/external/0267577771) Locked, Waiting on external entities >>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session 3 >>> (sofia/external/0267577771) Ended >>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close >>> Channel >>> sofia/external/0267577771 [CS_DESTROY] >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 >>> (sofia/external/0267577771) Callstate Change HANGUP -> DOWN >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 >>> (sofia/external/0267577771) Running State Change CS_DESTROY >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/external/0267577771) State DESTROY >>> 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 >>> sofia/external/0267577771 >>> SOFIA DESTROY >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 >>> sofia/external/0267577771 Standard DESTROY >>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/external/0267577771) State DESTROY going to sleep >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing >>> global variables >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [origination_caller_id_number]=[0264370536] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_from]=[0264370536] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_to]=[0249230704] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [plivo_app]=[true] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [originate_timeout]=[60] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>> [ignore_early_media]=[true] >>> 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel >>> sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] >>> 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 >>> (sofia/external/0249230704) State Change CS_NEW -> CS_INIT >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0249230704) Running State Change CS_INIT >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/external/0249230704) State INIT >>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 >>> sofia/external/0249230704 >>> SOFIA INIT >>> 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: >>> v=0 >>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>> s=FreeSWITCH >>> c=IN IP4 10.10.50.1 >>> t=0 0 >>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> a=sendrecv >>> >>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 >>> (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>> (sofia/external/0249230704) State INIT going to sleep >>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: >>> >>> ------------------------------------------------------------------------ >>> INVITE [hidden email]SIP/2.0 >>> >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>> Max-Forwards: 70 >>> From: "" <[hidden email] >>> >;tag=e6Be2N49HN2QN >>> To: <[hidden email] >>> > >>> >>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 INVITE >>> Contact: >>> User-Agent: >>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Allow-Events: talk, hold, conference, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 201 >>> X-FS-Support: update_display,send_info >>> Remote-Party-ID: >>> <[hidden email] >>> >;party=calling;screen=yes;privacy=off >>> >>> >>> v=0 >>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>> s=FreeSWITCH >>> c=IN IP4 10.10.50.1 >>> t=0 0 >>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>> a=rtpmap:101 telephone-event/8000 >>> a=fmtp:101 0-16 >>> a=ptime:20 >>> >>> ------------------------------------------------------------------------ >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0249230704) Running State Change CS_ROUTING >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/external/0249230704) State ROUTING >>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 >>> sofia/external/0249230704 >>> SOFIA ROUTING >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 >>> (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>> (sofia/external/0249230704) State ROUTING going to sleep >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>> (sofia/external/0249230704) State CONSUME_MEDIA >>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>> (sofia/external/0249230704) State CONSUME_MEDIA going to sleep >>> 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel >>> sofia/external/0249230704 entering state [calling][0] >>> recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 606 Not Acceptable >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>> From: "" <[hidden email] >>> >;tag=e6Be2N49HN2QN >>> To: <[hidden email] >>> >;tag=633680086 >>> >>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 INVITE >>> User-Agent: dble >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: >>> >>> ------------------------------------------------------------------------ >>> ACK [hidden email]SIP/2.0 >>> >>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>> Max-Forwards: 70 >>> From: "" <[hidden email] >>> >;tag=e6Be2N49HN2QN >>> To: <[hidden email] >>> >;tag=633680086 >>> >>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>> CSeq: 44398096 ACK >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------ >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel >>> sofia/external/0249230704 entering state [terminated][606] >>> 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup >>> sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal >>> sofia/external/0249230704 [KILL] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0249230704) Running State Change CS_HANGUP >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/external/0249230704) State HANGUP >>> 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel >>> sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 >>> sofia/external/0249230704 Standard HANGUP, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>> (sofia/external/0249230704) State HANGUP going to sleep >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 >>> (sofia/external/0249230704) Callstate Change DOWN -> HANGUP >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 >>> (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>> (sofia/external/0249230704) Running State Change CS_REPORTING >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>> (sofia/external/0249230704) State REPORTING >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 >>> sofia/external/0249230704 Standard REPORTING, cause: >>> INCOMPATIBLE_DESTINATION >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>> (sofia/external/0249230704) State REPORTING going to sleep >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 >>> (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send signal >>> sofia/external/0249230704 [BREAK] >>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 >>> (sofia/external/0249230704) Locked, Waiting on external entities >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 Originate >>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session 4 >>> (sofia/external/0249230704) Ended >>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close >>> Channel >>> sofia/external/0249230704 [CS_DESTROY] >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 >>> (sofia/external/0249230704) Callstate Change HANGUP -> DOWN >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 >>> (sofia/external/0249230704) Running State Change CS_DESTROY >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/external/0249230704) State DESTROY >>> 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 >>> sofia/external/0249230704 >>> SOFIA DESTROY >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 >>> sofia/external/0249230704 Standard DESTROY >>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>> (sofia/external/0249230704) State DESTROY going to sleep >>> >>> >>> >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> [hidden email] >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> [hidden email] >>> >>> 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: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> If you reply to this email, your message will be added to the >> discussion below: >> >> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591120.html >> To unsubscribe from SIP/2.0 606 Not Acceptable and >> INCOMPATIBLE_DESTINATION, click here. >> NAML >> > > > > -- > *Emmanuel O. Addai,* > *+233(0)26 757 7771* > > ------------------------------ > View this message in context: Re: SIP/2.0 606 Not Acceptable and > INCOMPATIBLE_DESTINATION > > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/d4e336d3/attachment-0001.html From drk at drkngs.net Mon May 27 23:00:01 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Mon, 27 May 2013 12:00:01 -0700 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: <111509FCE9EBF74BAB2CB38AAC521D9B23E34FE518@helle.dk.hq.firstcom.dk> Message-ID: <20130527190001.cd20d882@mail.tritonwest.net> Yes, I understand that. The method I gave you works for all cases, and you don't have to have a lot of special code for each case. It's much easier to have the logic once, and that way as your application grows, and needs to be changed, it's a lot easier. --Dave _____ From: Jon Sch?pzinsky [mailto:jos at firstcom.dk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 27 May 2013 11:29:10 -0700 Subject: Re: [Freeswitch-users] Splitting CDRs on call forward Hello Dave, The main problem isnt really the first scenario, as only the last leg needs to be billed. The biggest issue is with the second scenario, since both calls need to be billed, as theres two billing entities involved. I am also not using the FreeSWITCH directory, as the authentication and registration are handled by an upstream OpenSIPS instead. But thank you for the very thorough description. Its definitely inspiring working towards a solution. Right now im leaning towards using hooks, and taking care of the billing in lua, instead of using the internal CDR system. That way i can get it exactly as i want it :) /Jon ________________________________________ Fra: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] På vegne af Dave R. Kompel [drk at drkngs.net] Sendt: 27. maj 2013 18:58 Til: FreeSWITCH Users Help Emne: Re: [Freeswitch-users] Splitting CDRs on call forward This is a major can of worms. I have an comericial carrier softswitch based on FS, and what I did to be able to imlment billing the right legs to the right account was the following: Have a flag on your gateways so you can mark them as "pstn-uplink", or whatever you want to call it. This way you know that it's the "other" leg's endpoint is the entity you want to bill. Next, in your directory, make sure you set a variable for the user that has it's billing identity for example "myapp_user" or "myapp_account". If you use account then you can associate more then one user to the same account, if that's the way you're billing works. Now, if you make sure all calls to users, use the user/username at domain form of the dial string, you will also have the variable set on calls terminating to that user. Now you can bill, on normal calls both incoming calls (charge for DID usage) and outbound calls. When you get any inbound call from a gateway that's marked as "PSTN" then set a variable on that leg, when it calls a user, that indicates it's the PAID inbound leg for that user, before bridging to the "user" channel. To implment forwarding, in your processing of a call to a user, before bridging to the "user" channel, see if the users is forwarding, if so all you have to do, in your dialplan, or external code handling the dialplan is: "set_user to the user they called", then "transfer to: ${Forwarding_number} XML ${user_context}". Because you did the "set_user" first, the call will go back to CS_ROUTING, but look like it's from the user that did the forwarding. This way, if the call is from a gateway that isn't in the right context to dial the forwarding number, it won't matter, and the B-LEG will go through your dialplan logic, like the call is made form the user, and you can set a variable on the B_LEG in the dial string to tell it who the call is on behalf of, if it goes out a gateway that is marked as "PSTN". This way the same logic of how you tag the PSTN leg as being called from a user, is the same for a call made from the user, or forwared by the user. Since any users in the middle that are forwarding to internal numbers, it doesn't matter how many forwarding hops you go through, the original A-LEG if PSTN will get billed to the user it called, and the final B-LEG will get biled to the user that did the forwarding, that caused the call to leave your switch. --Dave ________________________________ From: Jon Sch?pzinsky [mailto:jos at firstcom.dk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Mon, 27 May 2013 09:01:49 -0700 Subject: [Freeswitch-users] Splitting CDRs on call forward Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Sch?pzinsky _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- This message has been scanned for viruses and dangerous content, and is believed to be clean. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130527/d2a6e27a/attachment.html From tculjaga at gmail.com Mon May 27 23:24:19 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 27 May 2013 21:24:19 +0200 Subject: [Freeswitch-users] bridging 2 call legs after a conference In-Reply-To: <20130527163156.f81d1b38@mail.tritonwest.net> References: <20130527163156.f81d1b38@mail.tritonwest.net> Message-ID: wow, thats great! .. thank you so much for the nice explanation :=) regards, T. On Mon, May 27, 2013 at 6:31 PM, Dave R. Kompel wrote: > ** > Getting the calls parked is the right way, but there is an easier way to > do it. The two legs that you want to remain in the call after the > conference bridge, you can just uuid_transfer them both to park as an > inline dialplan right in the transfer, then you're free to do the > uuid_bridge. > > The variable "park_after_bridge" is used in the core by > switch_ivr_bridge() funciton. It really only applies to a call that's been > bridged via that function, which would not apply in all cases. For example > if you called conference form the dialplan, or some other dp app that > doesn't use "switch_ivr_bridge". It also checks the variable on terminating > the bridge. This allows you to set the variable after the fact, while the > channel is bridged. > > However, since the call may of not started out bridged, the > park_after_bridge will not have any meaning. So just to be safe, and not > have to track every call via ESL, to know how it got to the state, I would > do a "uuid_transfer park inline" to each channel you want to remove > from the conference, and then you can do the "uuid_bridge". > > --Dave > > ------------------------------ > *From:* Tihomir Culjaga [mailto:tculjaga at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Mon, 27 May 2013 08:21:02 -0700 > *Subject:* Re: [Freeswitch-users] bridging 2 call legs after a conference > > > ya, i have park_after_bridge=true on the incoming call leg :=) ... thats > why it goes on park after leaving the conference... > > never thought park_after_bridge affects conference as well ... :) > > thanks anyway. > > T. > > > On Mon, May 27, 2013 at 4:13 PM, Tihomir Culjaga wrote: > >> hello folks, >> >> >> im controlling FS via ESL and what im trying to do is to make a 3-way >> conference and this works fine. What i have an issue with is when one >> conference member hangs up, i want to return back to a normal bridge >> between the 2 remaining members ... >> >> so what im doing is: >> >> On Incoming call from 1013 to FS >> answer the call: >> ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: >> execute","execute-app-name: answer"]) >> >> play an announcement to A leg: >> CMD uuid_broadcast "bgapi uuid_broadcast >> 00349e91-117e-4dcc-b2db-45d3a54c5fee 'greeting1.wav'" >> >> >> originate a new call and parks it: >> CMD originate "bgapi originate >> {origination_uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234,ignore_early_media=false,originate_timeout=15}user/1002 >> &park" >> >> >> bridge these 2 legs into a call: >> CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 >> 00349e91-117e-4dcc-b2db-45d3a54c5fee" >> >> >> >> originate a 2nd new call into a conference: >> CMD originate "bgapi originate >> {origination_uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f,ignore_early_media=false,originate_timeout=0}user/1014 >> &conference(supervizor36 at silent+flags{mute}) &park" >> >> >> join the 1st new call (user/1002) >> CMD_MULTI conference ["sendmsg >> fc3e44c5-63af-45bb-8a8e-8a35bac52234","call-command: >> execute","execute-app-name: conference","execute-app-arg: >> supervizor36 at silent"] >> >> join incoming call into the same conference >> CMD_MULTI conference ["sendmsg >> 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: >> execute","execute-app-name: conference","execute-app-arg: >> supervizor36 at silent"] >> >> >> unmute uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 >> CMD api "api conference supervizor36 unmute 20" >> >> >> Incoming call 00349e91-117e-4dcc-b2db-45d3a54c5fee must not hear >> uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 >> CMD api "api conference supervizor36 relate 21 20 nohear" >> >> >> so far its great! >> >> now, call uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f (the one originated >> into the conference) hangs up, 2 remaining calls (incoming call and 1st >> originated calls) can still hear each others... but i want to get rid of a >> conference since there are just 2 members hence no sense keeping the >> conference up... >> >> so i do: >> bridge the two remaining conference members together >> CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 >> 00349e91-117e-4dcc-b2db-45d3a54c5fee" >> >> and i get no audio!! >> >> >> >> looks like FS setting both channels on park ?? >> >> any idea ? >> >> >> >> 1. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1367 >> Session 30 (sofia/internal/sip:1014 at 192.168.254.112:5060) Ended >> 2. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1369 Close >> Channel sofia/internal/sip:1014 at 192.168.254.112:5060 [CS_DESTROY] >> 3. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:491 >> (sofia/internal/sip:1014 at 192.168.254.112:5060) Callstate Change >> HANGUP -> DOWN >> 4. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:494 >> (sofia/internal/sip:1014 at 192.168.254.112:5060) Running State Change >> CS_DESTROY >> 5. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY >> 6. 2013-05-27 15:34:13.724037 [DEBUG] mod_sofia.c:363 >> sofia/internal/sip:1014 at 192.168.254.112:5060 SOFIA DESTROY >> 7. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:86 >> sofia/internal/sip:1014 at 192.168.254.112:5060 Standard DESTROY >> 8. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 >> (sofia/internal/sip:1014 at 192.168.254.112:5060) State DESTROY going >> to sleep >> 9. >> 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1521 (sofia/internal/ >> sip:1002 at 192.168.254.116:5060) State Change CS_SOFT_EXECUTE -> >> CS_HIBERNATE >> 10. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 11. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1523 ( >> sofia/external/1013 at 192.168.254.93) State Change CS_EXCHANGE_MEDIA -> >> CS_HIBERNATE >> 12. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 13. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 14. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 15. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel >> leaving conference, cause: NONE >> 16. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel >> leaving conference, cause: NONE >> 17. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 18. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 >> sofia/internal/sip:1002 at 192.168.254.116:5060 Restore previous codec >> PCMA:8. >> 19. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 20. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 21. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 22. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 >> sofia/external/1013 at 192.168.254.93 Restore previous codec PCMA:8. >> 23. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 24. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 25. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1602 Write >> Lock ON >> 26. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/internal/sip:1002 at 192.168.254.116:5060] >> 27. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1605 Write >> Lock OFF >> 28. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 29. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 30. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 31. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:1366 ( >> sofia/internal/sip:1002 at 192.168.254.116:5060) State Change >> CS_HIBERNATE -> CS_RESET >> 32. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 33. >> 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/ >> sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE going to sleep >> 34. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 362 (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State >> Change CS_RESET >> 35. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 413 (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET >> 36. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:166 >> sofia/internal/sip:1002 at 192.168.254.116:5060 SOFIA RESET >> 37. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:721 >> sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM RESET >> 38. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:728 ( >> sofia/internal/sip:1002 at 192.168.254.116:5060) State Change CS_RESET >> -> CS_SOFT_EXECUTE >> 39. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 40. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 413 (sofia/internal/sip:1002 at 192.168.254.116:5060) State RESET going >> to sleep >> 41. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 362 (sofia/internal/sip:1002 at 192.168.254.116:5060) Running State >> Change CS_SOFT_EXECUTE >> 42. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 423 (sofia/internal/sip:1002 at 192.168.254.116:5060) State SOFT_EXECUTE >> 43. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:572 SOFIA >> SOFT_EXECUTE >> 44. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:746 >> sofia/internal/sip:1002 at 192.168.254.116:5060 CUSTOM SOFT_EXECUTE >> 45. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE >> THREAD DONE [sofia/external/1013 at 192.168.254.93] >> 46. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send >> signal sofia/internal/sip:1002 at 192.168.254.116:5060 [BREAK] >> 47. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr.c:2422 ( >> sofia/external/1013 at 192.168.254.93) State Change CS_HIBERNATE -> >> CS_PARK >> 48. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send >> signal sofia/external/1013 at 192.168.254.93 [BREAK] >> 49. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 420 (sofia/external/1013 at 192.168.254.93) State EXCHANGE_MEDIA going >> to sleep >> 50. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 362 (sofia/external/1013 at 192.168.254.93) Running State Change CS_PARK >> 51. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 426 (sofia/external/1013 at 192.168.254.93) State PARK >> 52. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c: >> 247 sofia/external/1013 at 192.168.254.93 Standard PARK >> 53. >> 54. >> 55. >> 56. freeswitch at internal> 2013-05-27 15:34:18.457165 [WARNING] >> sofia_reg.c:1400 SIP auth challenge (REGISTER) on sofia profile >> 'internal' for [1014 at 192.168.254.93]from ip 192.168.254.112 >> 57. >> 58. freeswitch at internal> >> 59. freeswitch at internal> >> 60. freeswitch at internal> show calls >> 61. >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch >> 62. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, >> 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013, >> 192.168.254.112,38515494471,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,cc01,Outbound >> Call,1002,,,,,,,,,,,,,,,,,,, >> 63. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, >> 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound >> Call,1002,,1002,1002 at 192.168.254.93 >> ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,cc01,1013,1013 >> ,,,,,,,,,,,,,,,,,,, >> 64. >> 65. 2 total. >> 66. >> 67. freeswitch at internal> show channels >> 68. >> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num >> 69. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, >> 1369661569,sofia/external/1013 at 192.168.254.93,CS_PARK,1013,1013, >> 192.168.254.112,38515494471,conference,supervizor36 at silent >> ,XML,default,PCMA,8000,64000,PCMA,8000,64000,,cc01,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,Outbound >> Call,1002 >> 70. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, >> 1369661585,sofia/internal/sip:1002 at 192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound >> Call,1002,,1002,conference,supervizor36 at silent,,default,PCMA,8000, >> 64000,PCMA,8000,64000,,cc01,1002 at 192.168.254.93 >> ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,1013,1013 >> 71. >> 72. 2 total. >> 73. >> 74. freeswitch at internal> >> 75. freeswitch at internal> >> 76. freeswitch at internal> >> 77. freeswitch at internal> 2013-05-27 15:34:57.732290 [DEBUG] >> switch_core_session.c:870 Send signal sofia/internal/sip:1002 at 192.168 >> .254.116:5060 [BREAK] >> >> >> >> >> >> Tihomir. >> >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130527/f679632b/attachment-0001.html From msc at freeswitch.org Tue May 28 01:09:02 2013 From: msc at freeswitch.org (Michael Collins) Date: Mon, 27 May 2013 14:09:02 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <8fee6197-5f94-418c-a298-f0004c469d95@email.android.com> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <8fee6197-5f94-418c-a298-f0004c469d95@email.android.com> Message-ID: http://wiki.freeswitch.org/wiki/Variable_dtmf_type Set that prior to the bridge. -MC On Sun, May 26, 2013 at 1:37 AM, Hermouet Erwan wrote: > Nobody know the good change to passe rfc2833 to info ? > > Tks advance > > > Michael Jerris a ?crit : > >> if you see the rtp events going out but you don't see it having any affect, try asking your provider? >> >> >> Mike >> >> On May 24, 2013, at 5:31 AM, ehermouet at bluetel.fr wrote: >> >> After some hours i foudn rtp event with wireshark. >>> >>> RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) >>> but no result on ivr outbound... >>> >>> >>> >>> >>> Le 2013-05-23 18:06, Michael Jerris a ?crit : >>> >>>> Have you looked at it to see if it is sending the dtmf? >>>> >>>> On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr wrote: >>>> >>>> you can found here my tcpdump file >>>>> >>>>> http://bluetelconnect.fr/tcpdump.log >>>>> >>>>> tks advance Michael >>>>> >>>>> >>>>> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>>>> >>>>>> you can use tcpdump to get a pcap. I didn't see anything wrong in >>>>>> review of what you have posted so far. >>>>>> >>>>>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr wrote: >>>>>> >>>>>> pcap. >>>>>>> >>>>>>> i send you the xml file and log in my previous email... because i >>>>>>> see >>>>>>> problem sometime... i'm sure i have error on my xml file. can you >>>>>>> check >>>>>>> it. ? >>>>>>> >>>>>>> tks >>>>>>> >>>>>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>>>> >>>>>>>> How do you use what? >>>>>>>> >>>>>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr wrote: >>>>>>>> >>>>>>>> how do you use it without interface ? it's server with only ssh >>>>>>>>> access. >>>>>>>>> tks >>>>>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>>>> >>>>>>>>>> this log does not seem to have a complete call let alone any >>>>>>>>>> attempt >>>>>>>>>> at dtmf. I don't see anything wrong from this log but as I >>>>>>>>>> said, >>>>>>>>>> its >>>>>>>>>> incomplete. If you pcap the traffic, do you see 2833 dtmf >>>>>>>>>> flowing >>>>>>>>>> ? >>>>>>>>>> >>>>>>>>>> Mike >>>>>>>>>> >>>>>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr wrote: >>>>>>>>>> >>>>>>>>>> Yes >>>>>>>>>>> >>>>>>>>>>> http://pastebin.freeswitch.org/20947 >>>>>>>>>>> >>>>>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>>>> >>>>>>>>>>>> Did you ever post a new log after you changed codec >>>>>>>>>>>> negotiation >>>>>>>>>>>> settings? >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> ------------------------------ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>> >>> ------------------------------ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> ------------------------------ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > Hermouet Erwan > Responsable technique > Bluetel > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130527/a2fd7df1/attachment.html From mbodbg at gmx.net Tue May 28 02:02:49 2013 From: mbodbg at gmx.net (mbo) Date: Tue, 28 May 2013 00:02:49 +0200 Subject: [Freeswitch-users] Getting CHANNEL_PROGRESS event only sometimes when originating call Message-ID: <43F67622-8165-4A57-97FA-6AED4D26F130@gmx.net> We are using Freeswitch version 1.2.9. connected with an Sangoma A104 card to the PSTN. If I originate a call though esl socket connection, I'm not always getting a CHANNEL_PROGRESS event. In the test scenario, I'm repeating the originate a couple of times with the same phone number and same parameters, sometimes I get CHANNEL_PROGRESS sometimes not. In the Q.931 trace I can see that I always get a ALERT message, which if I see it correctly, should be converted into a CHANNEL_PROGRESS event. We also have a demo system here with 2 Sangoma A101 cards connected with an crossed E1 cable (also Freeswitch 1.2.9). If I perform the same test here, I always get the CHANNEL_PROGRESS event. Are there any known issues that CHANNEL_PROGRESS event isn't fired reliable when originating calls via freetdm? Thanks Markus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/c7734494/attachment.html From xiaofengcanyuexp at 163.com Tue May 28 02:57:41 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Tue, 28 May 2013 06:57:41 +0800 Subject: [Freeswitch-users] sip_ph_X- header doesn't work for freetdm-sofiacall References: <201305280045030372439@163.com> Message-ID: <201305280657394843597@163.com> Anyone knows about this problem? It's appreciated. >-----Original Message----- >From: xiaofengcanyuexp [mailto:freeswitch-users at lists.freeswitch.org] >Sent: 2013-05-28 00:47:54 >Cc: >Subject: [Freeswitch-users] sip_ph_X- header doesn't work for freetdm-sofiacall >Dear support > >I got a problem to pass through SS7 info in SIP messages. Here is the code change and test call flow: >-------------------- >Code change and log: >-------------------- >1. At sangoma module:intercept ACM in ftmod_sangoma_ss7_*.c, parse the BCI and add it to the ftdmchan->call_data: > sngss7_add_var(sngss7_info, "ss7_acm_bci_hex", val); > >2. At freetdm module mod_freetdm.c, get the variable and set it to channel. Of which, the "prefix" is 'sip_ph_X-': > var_value = ftdm_sigmsg_get_var(sigmsg, "ss7_acm_bci_hex"); > if (!ftdm_strlen_zero(var_value)) > { > snprintf(variable_name, "%sTEST-BCI-HEX", prefix); > switch_channel_set_variable_printf(channel, variable_name, "%s", var_value); > } > >>From the test call log: > >2013-05-27 07:03:47.747361 [INFO] ftmod_sangoma_ss7_handle.c:362 [s1c1][1:1] [CIC:1]Rx ACM >2013-05-27 07:03:47.747361 [DEBUG] ftmod_sangoma_ss7_support.c:2920 [s1c1][1:1] ACM BCI parameter Hex: 0x1636 >2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5513 Get ACM BCI parameter from sangoma >2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5530 sip_ph_X-TEST-BCI-HEX:1636 >--------------------- >My purpose: >--------------------- >What I want to do is the encapsulate the sip_ph_X-TEST-BCI-HEX in the 183 SIP message. Like handling IAM in ftdm_channel_from_event(), >once it sets the the variable like below, the "X-FreeTDM-SpanNumber" will automatically attached in INVITE message. > switch_channel_set_variable_printf(channel, "sip_h_X-FreeTDM-SpanNumber", "%d", spanid); > >--------------------- >Problems: >--------------------- >While, what's strange are: >1. I add one line in dialplan/default.xml to log the variable with: > > The freeswitch.log show ${sip_ph_X-TRUSTID-BCI-HEX} is empty >2. The 183 message does not attach the 'X-TRUSTID-BCI-HEX'. > >Can someone help to figure out what's wrong with it? > >Thanks >------------------- >2013-05-28 > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 = = = = = = = = = = = = = = = = = = = = From nandy1925 at gmail.com Tue May 28 06:34:27 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Tue, 28 May 2013 10:34:27 +0800 Subject: [Freeswitch-users] Best Installation MOD_GSMOPEN In-Reply-To: References: Message-ID: After wresting the problem installing gsmopen on Centos 6.4 x_64 (** libctb-2.6.so problem), I feel this information should be added in the Wiki in the pre-requisites ldconfig <-- after installing gsm library ---- Add this --- Check if the new library is included # ldconfig -p | grep gsm If no entries found, correct this problem by using the ldconfig examples at http://linux.101hacks.com/unix/ldconfig/ Run ldconfig again ----- End --- In my case, I added this rule file in /etc/ld.so.d directory # local-libs.conf /usr/local/lib /nandy On Tue, Feb 26, 2013 at 2:10 AM, Josue Diaz Cruz wrote: > ** > Good Afternoon. > > I beg your pardon if my question is really from a newbee. > > I am trying to install gsmopen in a system. I did it with CentOS 6 X64, > Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. > Cause i allways have problems in one point of the installation. Depending > the distro could it be one or other. I was thinking that could be nice to > give a complete requierements (libs) to install before do anything. > > If some one can provide me a easy, simple and not troubles, could be nice: > > * System and version > * List of requierements. > * Better steps to avoid issues with libraries and compile. > > What i tried: > > Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when > try to load mod_gsmopen.so) > Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with > the ttyUSB port) > Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with > the ttyUSB port) > > E169 works perfectly. I have a mobile partner with voicecall and i test it > calling my land line without issues. > > > Thank you very much for your consideration. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/e3441b30/attachment-0001.html From ehermouet at bluetel.fr Tue May 28 11:07:38 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Tue, 28 May 2013 09:07:38 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <8fee6197-5f94-418c-a298-f0004c469d95@email.android.com> Message-ID: <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Le 2013-05-27 23:09, Michael Collins a ?crit?: > http://wiki.freeswitch.org/wiki/Variable_dtmf_type [48] > > Set that prior to the bridge. > -MC > > On Sun, May 26, 2013 at 1:37 AM, Hermouet Erwan [49]> wrote: > >> Nobody know the good change to passe rfc2833 to info ? >> >> Tks advance >> >> Michael Jerris a ?crit?: >> >>> if you see the rtp events going out but you dont see it having any >>> affect, try asking your provider? >>> >>> Mike >>> >>> On May 24, 2013, at 5:31 AM, ehermouet at bluetel.fr wrote: >>> >>>> After some hours i foudn rtp event with wireshark. >>>> >>>> RTP EVENT 60 Payload type=RTP Event, DTMF Five 5 (end) >>>> but no result on ivr outbound... >>>> >>>> Le 2013-05-23 18:06, Michael Jerris a ?crit : >>>> >>>>> Have you looked at it to see if it is sending the dtmf? >>>>> >>>>> On May 23, 2013, at 11:37 AM, ehermouet at bluetel.fr [6] wrote: >>>>> >>>>>> you can found here my tcpdump file >>>>>> >>>>>> http://bluetelconnect.fr/tcpdump.log [5] >>>>>> >>>>>> tks advance Michael >>>>>> >>>>>> Le 2013-05-23 16:58, Michael Jerris a ?crit : >>>>>> >>>>>>> you can use tcpdump to get a pcap. I didnt see anything >>>>>>> wrong in >>>>>>> review of what you have posted so far. >>>>>>> >>>>>>> On May 23, 2013, at 10:05 AM, ehermouet at bluetel.fr [4] >>>>>>> wrote: >>>>>>> >>>>>>>> pcap. >>>>>>>> >>>>>>>> i send you the xml file and log in my previous email... >>>>>>>> because i >>>>>>>> see >>>>>>>> problem sometime... im sure i have error on my xml file. >>>>>>>> can you >>>>>>>> check >>>>>>>> it. ? >>>>>>>> >>>>>>>> tks >>>>>>>> >>>>>>>> Le 2013-05-23 15:24, Michael Jerris a ?crit : >>>>>>>> >>>>>>>>> How do you use what? >>>>>>>>> >>>>>>>>> On May 23, 2013, at 9:07 AM, ehermouet at bluetel.fr [3] >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>>> how do you use it without interface ? its server >>>>>>>>>> with only ssh >>>>>>>>>> access. >>>>>>>>>> tks >>>>>>>>>> Le 2013-05-23 14:58, Michael Jerris a ?crit : >>>>>>>>>> >>>>>>>>>>> this log does not seem to have a complete call let >>>>>>>>>>> alone any >>>>>>>>>>> attempt >>>>>>>>>>> at dtmf. I dont see anything wrong from this log >>>>>>>>>>> but as I >>>>>>>>>>> said, >>>>>>>>>>> its >>>>>>>>>>> incomplete. If you pcap the traffic, do you see >>>>>>>>>>> 2833 dtmf >>>>>>>>>>> flowing >>>>>>>>>>> ? >>>>>>>>>>> >>>>>>>>>>> Mike >>>>>>>>>>> >>>>>>>>>>> On May 23, 2013, at 8:44 AM, ehermouet at bluetel.fr >>>>>>>>>>> [2] wrote: >>>>>>>>>>> >>>>>>>>>>>> Yes >>>>>>>>>>>> >>>>>>>>>>>> http://pastebin.freeswitch.org/20947 [1] >>>>>>>>>>>> >>>>>>>>>>>> Le 2013-05-23 14:28, Michael Jerris a ?crit : >>>>>>>>>>>> >>>>>>>>>>>>> Did you ever post a new log after you changed >>>>>>>>>>>>> codec >>>>>>>>>>>>> negotiation >>>>>>>>>>>>> settings? >>>>> >>>>> ------------------------- >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org [7] >>>>> http://www.freeswitchsolutions.com [8] >>>>> >>>>> >>>>> [9] >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org [10] >>>>> http://wiki.freeswitch.org [11] >>>>> http://www.cluecon.com [12] >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org [13] >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> [14] >>>>> >>>>> >>>> >>> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> [15] >>>>> http://www.freeswitch.org [16] >>>> >>>> ------------------------- >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org [17] >>>> http://www.freeswitchsolutions.com [18] >>>> >>>> >>>> [19] >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org [20] >>>> http://wiki.freeswitch.org [21] >>>> http://www.cluecon.com [22] >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org [23] >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> [24] >>>> >>> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> [25] >>>> http://www.freeswitch.org [26] >>> >>> ------------------------- >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org [27] >>> http://www.freeswitchsolutions.com [28] >>> >>> >>> [29] >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org [30] >>> http://wiki.freeswitch.org [31] >>> http://www.cluecon.com [32] >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org [33] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [34] >>> >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> [35] >>> http://www.freeswitch.org [36] >> >> Hermouet Erwan >> Responsable technique >> Bluetel >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [38] >> http://www.freeswitchsolutions.com [39] >> >> >> [40] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [41] >> http://wiki.freeswitch.org [42] >> http://www.cluecon.com [43] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [44] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [45] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [46] >> http://www.freeswitch.org [47] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [50] > http://www.ClueCon.com [51] > http://www.OSTAG.org [52] > > > > Links: > ------ > [1] http://pastebin.freeswitch.org/20947 > [2] mailto:ehermouet at bluetel.fr > [3] mailto:ehermouet at bluetel.fr > [4] mailto:ehermouet at bluetel.fr > [5] http://bluetelconnect.fr/tcpdump.log > [6] mailto:ehermouet at bluetel.fr > [7] mailto:consulting at freeswitch.org > [8] http://www.freeswitchsolutions.com > [9] > [10] http://www.freeswitch.org > [11] http://wiki.freeswitch.org > [12] http://www.cluecon.com > [13] mailto:FreeSWITCH-users at lists.freeswitch.org > [14] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [15] http://lists.freeswitch.org/mailman/options/freeswitch-users > [16] http://www.freeswitch.org > [17] mailto:consulting at freeswitch.org > [18] http://www.freeswitchsolutions.com > [19] > [20] http://www.freeswitch.org > [21] http://wiki.freeswitch.org > [22] http://www.cluecon.com > [23] mailto:FreeSWITCH-users at lists.freeswitch.org > [24] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [25] http://lists.freeswitch.org/mailman/options/freeswitch-users > [26] http://www.freeswitch.org > [27] mailto:consulting at freeswitch.org > [28] http://www.freeswitchsolutions.com > [29] > [30] http://www.freeswitch.org > [31] http://wiki.freeswitch.org > [32] http://www.cluecon.com > [33] mailto:FreeSWITCH-users at lists.freeswitch.org > [34] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [35] http://lists.freeswitch.org/mailman/options/freeswitch-users > [36] http://www.freeswitch.org > [37] mailto:mike at jerris.com > [38] mailto:consulting at freeswitch.org > [39] http://www.freeswitchsolutions.com > [40] > [41] http://www.freeswitch.org > [42] http://wiki.freeswitch.org > [43] http://www.cluecon.com > [44] mailto:FreeSWITCH-users at lists.freeswitch.org > [45] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [46] http://lists.freeswitch.org/mailman/options/freeswitch-users > [47] http://www.freeswitch.org > [48] http://wiki.freeswitch.org/wiki/Variable_dtmf_type > [49] mailto:ehermouet at bluetel.fr > [50] http://www.FreeSWITCH.org > [51] http://www.ClueCon.com > [52] http://www.OSTAG.org From mehroz.ashraf85 at gmail.com Tue May 28 11:26:16 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 28 May 2013 00:26:16 -0700 (PDT) Subject: [Freeswitch-users] SSL/TLS customized encryption. In-Reply-To: <1364818573173-7589290.post@n2.nabble.com> References: <1364394004803-7589146.post@n2.nabble.com> <1364476282723-7589184.post@n2.nabble.com> <1364539539133-7589238.post@n2.nabble.com> <1364818573173-7589290.post@n2.nabble.com> Message-ID: <1369725976961-7591164.post@n2.nabble.com> HI all, As far i have learnt, openssl should support the required encryption model. lets say, now i have got the version of openSSL meeting the standards. I want to restrict SIP/TLS cipher to support only the said encryption. Question: where do i have to mention , to use only that specific cipher model . (like, if its HTTPS, i would mention it somewhere in apache settings or nginx settings). Does FS support it or ill have to play directly with openssl? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption-tp7589146p7591164.html Sent from the freeswitch-users mailing list archive at Nabble.com. From regis.freeswitch.org at tornad.net Tue May 28 12:37:01 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 28 May 2013 10:37:01 +0200 Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: <1369661283368-7591135.post@n2.nabble.com> References: <1369661283368-7591135.post@n2.nabble.com> Message-ID: mod_callcenter reads configuration only once from xml_curl. You have to do a queue reload each time you change the agent/tier configuration, or manualy doing the modification with callcenter_config on cli. It's not like dialplan or directory configuration which doing a http request on each use. 2013/5/27 Anatolii > Hello all! > > Freeswitch version - 1.2.8, installed in Centos. > > Freeswitch always goes for configuration to our server (on which CakePHP > is situated) > > I have problem with callcenter module in freeswitch. > > In mod_xml_curl add two bindings: dialplan and configuration. > > < binding name="configuration"> > > < param name="use-dynamic-url" value="yes"/> > > < param name="gateway-url" value=" > http://example.comt/freeswitch/configuration/${our_queue_name}${company_id}" > bindings="configuration"/> > > < /binding> > > So, when I restart (start) freeswitch or reload mod_callcenter - FS > request default configuration for callcenter and it works great (FS add > queues, agents and tiers). > > After that in dialplan for specific number I set application callcenter > with specific name and set 2 global variables, *our_queue_name* and * > company_id*, which send to me with a name of queue and company id. -- > this works fine > > When I receive from freeswitch request - I configure xml from db data and > send FS callcenter configuration. -- works fine. *FS get correct xml, but > add only queue settings and absolutely ignoring adding agents and tiers.*-- This is my problem for now. > > I try to solve this situation by using lua scripts - no results. > > > ------------------------------ > View this message in context: Problem with callcenter configuration > (don't register agents and tiers) > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/4f35a977/attachment.html From adahary at gmail.com Tue May 28 12:39:01 2013 From: adahary at gmail.com (adahary) Date: Tue, 28 May 2013 01:39:01 -0700 (PDT) Subject: [Freeswitch-users] FS bypass ZRTP with TURN relay Message-ID: <1369730341924-7591165.post@n2.nabble.com> I've setup the FS with bypass_media mode to let endpoints to pass media through a TURN server. The dialplan is set dynamically to bypass_mode=true. The endpoints can now register SIP with FS and then pass media between thenself via the TURN relay (FS not in between) with no one-way-audio issue. The problem is when I setup FS and the endpoints with ZRTP. Even though all RTPs are passing directly between endpoints and FS log indications show that ZRTP is activated on both endpoints (including a=zrtp-hash: in both SDPs) and still the ZRTP is not poping up in the endpoints display. I assume that once media is being relayed between endpoints then ZRTP should pass between like it happens when both endpoints on the same LAN network. What could cause the ZRTP not to work between endpoints when media is bypassing the FS? regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-relay-tp7591165.html Sent from the freeswitch-users mailing list archive at Nabble.com. From regis.freeswitch.org at tornad.net Tue May 28 12:38:40 2013 From: regis.freeswitch.org at tornad.net (Regis M) Date: Tue, 28 May 2013 10:38:40 +0200 Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: References: <1369661283368-7591135.post@n2.nabble.com> Message-ID: Humm, sorry, I think I did a missread... your problem is a step backward... it's work great for us, we send the agent and tier info throw http as it was an xml callcenter configuration 2013/5/28 Regis M > mod_callcenter reads configuration only once from xml_curl. > You have to do a queue reload each time you change the agent/tier > configuration, or manualy doing the modification with callcenter_config on > cli. > > It's not like dialplan or directory configuration which doing a http > request on each use. > > > > > 2013/5/27 Anatolii > >> Hello all! >> >> Freeswitch version - 1.2.8, installed in Centos. >> >> Freeswitch always goes for configuration to our server (on which CakePHP >> is situated) >> >> I have problem with callcenter module in freeswitch. >> >> In mod_xml_curl add two bindings: dialplan and configuration. >> >> < binding name="configuration"> >> >> < param name="use-dynamic-url" value="yes"/> >> >> < param name="gateway-url" value=" >> http://example.comt/freeswitch/configuration/${our_queue_name}${company_id}" >> bindings="configuration"/> >> >> < /binding> >> >> So, when I restart (start) freeswitch or reload mod_callcenter - FS >> request default configuration for callcenter and it works great (FS add >> queues, agents and tiers). >> >> After that in dialplan for specific number I set application callcenter >> with specific name and set 2 global variables, *our_queue_name* and * >> company_id*, which send to me with a name of queue and company id. -- >> this works fine >> >> When I receive from freeswitch request - I configure xml from db data and >> send FS callcenter configuration. -- works fine. *FS get correct xml, >> but add only queue settings and absolutely ignoring adding agents and tiers. >> * -- This is my problem for now. >> >> I try to solve this situation by using lua scripts - no results. >> >> >> ------------------------------ >> View this message in context: Problem with callcenter configuration >> (don't register agents and tiers) >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130528/bea13976/attachment-0001.html From sertys at gmail.com Tue May 28 14:01:52 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 28 May 2013 12:01:52 +0200 Subject: [Freeswitch-users] FS bypass ZRTP with TURN relay In-Reply-To: <1369730341924-7591165.post@n2.nabble.com> References: <1369730341924-7591165.post@n2.nabble.com> Message-ID: Check in you profile that you have Otherwise zrtp won't work on some UAs(like the pjsip based) even though the zrtp-hash is passed along. On Tue, May 28, 2013 at 10:39 AM, adahary wrote: > I've setup the FS with bypass_media mode to let endpoints to pass media > through a TURN server. > The dialplan is set dynamically to bypass_mode=true. > The endpoints can now register SIP with FS and then pass media between > thenself via the TURN relay (FS not in between) with no one-way-audio > issue. > > The problem is when I setup FS and the endpoints with ZRTP. > Even though all RTPs are passing directly between endpoints and FS log > indications show that ZRTP is activated on both endpoints (including > a=zrtp-hash: in both SDPs) and still the ZRTP is not poping up in the > endpoints display. > I assume that once media is being relayed between endpoints then ZRTP > should > pass between like it happens when both endpoints on the same LAN network. > > What could cause the ZRTP not to work between endpoints when media is > bypassing the FS? > > regards > > Assaf > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-relay-tp7591165.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://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/20130528/fe6b2a5a/attachment.html From adahary at gmail.com Tue May 28 14:36:51 2013 From: adahary at gmail.com (adahary) Date: Tue, 28 May 2013 03:36:51 -0700 (PDT) Subject: [Freeswitch-users] FS bypass ZRTP with TURN relay In-Reply-To: References: <1369730341924-7591165.post@n2.nabble.com> Message-ID: <1369737411348-7591169.post@n2.nabble.com> Daniel, It is already set to: . ZRTP is working end-2-end in bypass_mode when the two endpoints are on the same LAN network. It is just NOT working when the enpoints are on different networks and being relayed via a TURN server (endpoints setup with STUN/TURN/ICE). I can see that the FS handing over the SDPs between the endpoints with endpoints direct relay ip:port and the zrtp hash and also the FS log indicated that the ZRTP is passthru a<->b BUT somehow the ZRTP doesn't happen on both endpoints! Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-relay-tp7591165p7591169.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Tue May 28 16:30:36 2013 From: mike at jerris.com (Michael Jerris) Date: Tue, 28 May 2013 08:30:36 -0400 Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> <8D02713727B38A2-18C4-18EFB@webmail-m236.sysops.aol.com> <8D0273532DCA6A6-18C4-19677@webmail-m236.sysops.aol.com> Message-ID: <9A1EED9C-199C-44BB-9948-208E7DB6891D@jerris.com> This sort of snark is totally uncalled for on the mailing list. If you can't be thankful for the people helping you for free, maybe its best you go find help elsewhere. Mike On May 25, 2013, at 4:01 AM, John M wrote: > Hi Daniel, > > Thanks for your description, it is much appreciated. :-) > > 5 word one liners from people too lazy to explain properly would really be best if they didn't reply at all. > > Cheers, thanks again. > > -Jm > > > > -----Original Message----- > From: Daniel Ivanov > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 5:57 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > It is true that the luasql driver is overly basic and poorly documented . Unfortunately mysql doesn't support RETURNING clause like pgsql and oracle. You should however try SELECT LAST_INSERT_ID(); right after the insert query. I cannot guarantee it works due to the unknown nature(to me that is) of the luasql transaction handling, but it should keep a transaction open for as long as a db handler lives. > On May 25, 2013 7:03 AM, "John M" wrote: > Hi Seven Du, > > I'd really like to know if this is possible too, couldn't find it documented anywhere. > > Instead of being cryptic, if you know the answer won't you please help by explaining what the RETURNING clause is and how to use it? > > Does it somehow return mysql_insert_id()? > > How should we use it. > > You help is invaluable and is contributing to the freeswitch community. > > -Jm > > > -----Original Message----- > From: Seven Du > To: FreeSWITCH Users Help > Sent: Sat, May 25, 2013 12:52 pm > Subject: Re: [Freeswitch-users] dbh:query - insert id > > Maybe try the RETURNING clause ? > > -- > Seven Du > http://www.freeswitch.org.cn > http://about.me/dujinfang > http://www.dujinfang.com > > Sent with Sparrow > > On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: >> Hello All >> >> How to get the id value after insert a record a record using dbh:query >> >> table_a - columns. >> >> id - auto increment >> field1 >> field2 >> >> >> dbh:query("insert into table_a ( field1,field2) values ('11','Test')") >> >> >> After insert how to get the table_a - id value for the inserted record? >> >> Thanks >> Lloyd >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/a5f9bd93/attachment-0001.html From sertys at gmail.com Tue May 28 16:34:14 2013 From: sertys at gmail.com (Daniel Ivanov) Date: Tue, 28 May 2013 14:34:14 +0200 Subject: [Freeswitch-users] FS bypass ZRTP with TURN relay In-Reply-To: <1369737411348-7591169.post@n2.nabble.com> References: <1369730341924-7591165.post@n2.nabble.com> <1369737411348-7591169.post@n2.nabble.com> Message-ID: What are the useragents for the legs. Ive never used the TURN relay, but would suggest looking there if it's doing some handling. Or also look at the verbose logs on the UAs. On May 28, 2013 1:40 PM, "adahary" wrote: > Daniel, > > It is already set to: . > > ZRTP is working end-2-end in bypass_mode when the two endpoints are on the > same LAN network. > It is just NOT working when the enpoints are on different networks and > being > relayed via a TURN server (endpoints setup with STUN/TURN/ICE). > I can see that the FS handing over the SDPs between the endpoints with > endpoints direct relay ip:port and the zrtp hash and also the FS log > indicated that the ZRTP is passthru a<->b BUT somehow the ZRTP doesn't > happen on both endpoints! > > Assaf > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-relay-tp7591165p7591169.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://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/20130528/aac2a0b2/attachment.html From mehroz.ashraf85 at gmail.com Tue May 28 16:59:00 2013 From: mehroz.ashraf85 at gmail.com (mehroz) Date: Tue, 28 May 2013 05:59:00 -0700 (PDT) Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <8D026AB5F145D0E-18C4-15C15@webmail-m236.sysops.aol.com> References: <1369386502709-7591047.post@n2.nabble.com> <1369395564138-7591052.post@n2.nabble.com> <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> <1369407298402-7591066.post@n2.nabble.com> <8D026AB5F145D0E-18C4-15C15@webmail-m236.sysops.aol.com> Message-ID: <1369745940420-7591172.post@n2.nabble.com> yes, something like that you mention. But the DB schema and script are imported from Freeswitch-contrib, and it says to define a domain name, and assign an ID to it. and then while declaring directory, associate them to that domain ID. i dont see any way of associated a user with 2 IDs. But, i somehow managaed to do that, i.e, disabling/commenting -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7591047p7591172.html Sent from the freeswitch-users mailing list archive at Nabble.com. From adahary at gmail.com Tue May 28 17:06:46 2013 From: adahary at gmail.com (adahary) Date: Tue, 28 May 2013 06:06:46 -0700 (PDT) Subject: [Freeswitch-users] FS bypass ZRTP with TURN relay In-Reply-To: References: <1369730341924-7591165.post@n2.nabble.com> <1369737411348-7591169.post@n2.nabble.com> Message-ID: <1369746406119-7591173.post@n2.nabble.com> I'm using Android CSimpleSip for both endpoints. I also suspect that the problem may be related to the CSimpleSip client but I do not familer with another Win/Linux client that supports both TURN and ZRTP. If anyone knows then please update me. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-relay-tp7591165p7591173.html Sent from the freeswitch-users mailing list archive at Nabble.com. From intralanman at freeswitch.org Tue May 28 17:42:18 2013 From: intralanman at freeswitch.org (Raymond Chandler) Date: Tue, 28 May 2013 09:42:18 -0400 Subject: [Freeswitch-users] Centralized SIP directory In-Reply-To: <1369745940420-7591172.post@n2.nabble.com> References: <1369386502709-7591047.post@n2.nabble.com> <1369395564138-7591052.post@n2.nabble.com> <8D026A1FF5A40EE-18C4-153A8@webmail-m236.sysops.aol.com> <1369407298402-7591066.post@n2.nabble.com> <8D026AB5F145D0E-18C4-15C15@webmail-m236.sysops.aol.com> <1369745940420-7591172.post@n2.nabble.com> Message-ID: <51A4B43A.2020206@freeswitch.org> On 13-05-28 08:59 AM, mehroz wrote: > yes, something like that you mention. But the DB schema and script are > imported from Freeswitch-contrib, and it says to define a domain name, and > assign an ID to it. and then while declaring directory, associate them to > that domain ID. i dont see any way of associated a user with 2 IDs. If you don't actually need the ability to support multiple domains, you could set the force params on each box so the domain is always the same. If you set the following params with "your.domain.tld" instead of "$${domain}", then it would force the domain to be your.domain.tld no matter which box was requesting the config. you can see the wiki for specifics on the following params. -Ray From cal.leeming at simplicitymedialtd.co.uk Tue May 28 17:54:25 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 28 May 2013 14:54:25 +0100 Subject: [Freeswitch-users] Splitting CDRs on call forward In-Reply-To: References: Message-ID: Hi Jon, I've just finished part #1 of my write up on ways to make multi-tenant SIP testing less tedious. Have a read over at; http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing If you haven't got time to read, just take a look at this instead; http://i.imgur.com/e9kUdxg.jpg Hope this helps Cal On Mon, May 27, 2013 at 5:01 PM, Jon Sch?pzinsky wrote: > Hi List, > > I am implementing call forwarding on a multi tenant system, and therefore > need to split CDR's when the call forward happens, so that if the > receiving user also has his account call forwarded, he pays for his part > of the call. > > A calls B > B forwards to C > C forwards to an external mobile phone. > > B has a free call from B to C, but C needs to pay for the forwarding to > the mobile phone. Therefore i need a separate CDR for the C to Mobile > phone call. > > Another example would be this > > A works in Company A, and B works in Company B > They are both users on our system, and therefore is on the same freeswitch. > > A calls B > B Forwards to an external mobile phone. > > Here A needs to pay for the call from A to B, and B needs to pay for the > call being forwarded to his mobile phone. > > Do anybody have an idea as to how to implement this in freeswitch. Back in > my Asterisk days, this would be done by the ForkCDR command. > > > Venlig hilsen/kind regards > > Jon Leren Sch?pzinsky > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/06acfb9d/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue May 28 18:03:46 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 28 May 2013 15:03:46 +0100 Subject: [Freeswitch-users] Better approach for multi-tenant SIP testing (part 1) Message-ID: Hello all, After nearly a year of active development on building our own voice platform, we've adopted some cool tricks to speed up the workflow and make testing less tedious. Over the next few months I'll be posting more articles on the tricks we have used and the lessons we have learnt. Here is part 1 - a write up about a neat hack we used to virtualize phone testing; http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing For those that don't have time to read, here is a screenshot; http://i.imgur.com/e9kUdxg.jpg This trick, which we've used in development for approx 6 months, has saved countless hours in development overheads, with one-click start/stop functionality and all the necessary modules right there in one place. Of course, this should never be used as a full replacement as nothing beats a proper hardware test to catch those weird edge cases, but it does allow you to do the majority of your testing (95%) from a single screen rather than a desk full of phones. It also makes attaching wireshark a lot easier, without having to use span ports and the alike. Hope this helps someone else! Cal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/f99a219b/attachment.html From ashwinrath at gmail.com Tue May 28 18:12:37 2013 From: ashwinrath at gmail.com (Ashwin Rath) Date: Tue, 28 May 2013 19:42:37 +0530 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs Message-ID: Hi I am trying to configure FS as a B2BUA which can do ptime conversions between the ingress and egress leg. The question is, if both legs have G729 as the codec but ptime 20 on one and 30 on another, would i still need to get the licensed mod_com_g729 or can it work with the passthrough mod_g729. I already tried it out, but it doesnt seem to work. Is there some configuration needed for this ? -- Thegrid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/a640d5bf/attachment-0001.html From anthony.minessale at gmail.com Tue May 28 18:51:30 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 28 May 2013 09:51:30 -0500 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: References: Message-ID: Set the variable passthru_ptime_mismatch=true on both legs of the call. That's the best we can do. If that doesn't work you'll need licenses. On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath wrote: > Hi > > I am trying to configure FS as a B2BUA which can do ptime conversions > between the ingress and egress leg. > > The question is, if both legs have G729 as the codec but ptime 20 on one > and 30 on another, would i still need to get the licensed mod_com_g729 or > can it work with the passthrough mod_g729. I already tried it out, but it > doesnt seem to work. > > Is there some configuration needed for this ? > > -- > Thegrid > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/29af4725/attachment.html From j_mj at aol.com Tue May 28 19:00:17 2013 From: j_mj at aol.com (John M) Date: Tue, 28 May 2013 11:00:17 -0400 (EDT) Subject: [Freeswitch-users] dbh:query - insert id In-Reply-To: <1F177F3A96B54D738071A078F0B60576@gmail.com> References: <1F177F3A96B54D738071A078F0B60576@gmail.com> Message-ID: <8D029CB32F74E42-18C4-2B2D8@webmail-m236.sysops.aol.com> I'd like to offer an apology to you Seven and to the list in general for an inappropriate comment I made about your offer of free assistance regarding the dbh insert id. My comment was rude and uncalled for and I will behave more appropriately in the future. The mailing list is an invaluable source of help and any offers of assistance large or small should be accepted graciously. I hope you will accept this apology. Humbly, sincerely, John -----Original Message----- From: Seven Du To: FreeSWITCH Users Help Sent: Sat, May 25, 2013 12:52 pm Subject: Re: [Freeswitch-users] dbh:query - insert id Maybe try the RETURNING clause ? -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow On Saturday, May 25, 2013 at 8:14 AM, Lloyd Aloysius wrote: Hello All How to get the id value after insert a record a record using dbh:query table_a - columns. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130528/69b5477a/attachment.html From msc at freeswitch.org Tue May 28 19:22:43 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 08:22:43 -0700 Subject: [Freeswitch-users] add-users scripts In-Reply-To: References: Message-ID: On Sun, May 26, 2013 at 9:55 PM, Adam Cooley wrote: > Thanks brian, > > I'm just wondering where I get the add_users script referenced in the > freeswitch cookbook and others? Also finding with a fresh install of > freeswitch 1.2.9 with centos and yum, non of the .wav files are loaded for > voicemail. Where do I get those files from? > Hi Adam, I wrote the add_users script and the recipe you mentioned in the cookbook The script you seek is in ${fs_source}/scripts/perl Just run it with --help and it'll spit out a bunch of info for you. -MC > -adam > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/80970f37/attachment.html From cesar.bermudez at gmail.com Tue May 28 20:01:06 2013 From: cesar.bermudez at gmail.com (Cesar Bermudez) Date: Tue, 28 May 2013 10:01:06 -0600 Subject: [Freeswitch-users] Better approach for multi-tenant SIP testing (part 1) In-Reply-To: References: Message-ID: Nice Work !!! , when the second part come? Regards On Tue, May 28, 2013 at 8:03 AM, Cal Leeming [Simplicity Media Ltd] < cal.leeming at simplicitymedialtd.co.uk> wrote: > Hello all, > > After nearly a year of active development on building our own voice > platform, we've adopted some cool tricks to speed up the workflow and make > testing less tedious. > > Over the next few months I'll be posting more articles on the tricks we > have used and the lessons we have learnt. > > Here is part 1 - a write up about a neat hack we used to virtualize phone > testing; > > http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing > > For those that don't have time to read, here is a screenshot; > http://i.imgur.com/e9kUdxg.jpg > > This trick, which we've used in development for approx 6 months, has saved > countless hours in development overheads, with one-click start/stop > functionality and all the necessary modules right there in one place. > > Of course, this should never be used as a full replacement as nothing > beats a proper hardware test to catch those weird edge cases, but it does > allow you to do the majority of your testing (95%) from a single screen > rather than a desk full of phones. It also makes attaching wireshark a lot > easier, without having to use span ports and the alike. > > Hope this helps someone else! > > Cal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/53c816c6/attachment-0001.html From msc at freeswitch.org Tue May 28 20:16:09 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 09:16:09 -0700 Subject: [Freeswitch-users] ESL using bridge app doesn't return which gateway was used In-Reply-To: References: Message-ID: On Thu, May 23, 2013 at 4:22 PM, Clinton Goudie-Nice wrote: > When you make a bridge command using esl, where you specify multiple > gateways or sip dials separated by or bars, you can't figure out which > gateway was used. > > For example, if you bridge to something like this: > > sofia/gateway/SBC-GW2/+18019600000|sofia/gateway/SBC-GW1/+18019600000 > Some have used this technique: [mygw=gw2]sofia/gateway/SBC-GW2/+18019600000 |[mygw=gw1]sofia/gateway/SBC-GW1/+18019600000 After the bridge, the chan var mygw will contain the value "gw2" or "gw1" depending on which dialstring connected the call. -MC The call could be bridged to either GW2 or GW1. > > When the CHANNEL_BRIDGE event is returned, you can see the original string > in variable_current_application_data, and you may be able to infer the > destination based on IP address, but nothing clearly says what gateway is > used. > > If you turn on the all events firehose, you can see the CHANNEL_CREATE > event come over the socket, and it does contain variable_sip_gateway_name > with the actual name of the gateway, however I can't devise a way to access > that data using the org.freeswitch.esl.client library, and even if I could, > I still don't want all events for this system. > > Is it possible to get this information returned in any meaningful way > through the ESL layer, either by an api command to query, or the setting of > a variable that will give me back which gateway a bridge was performed > through? > > If none of that is possible, this sounds worthy of filing a bug to return > the variable_sip_gateway_name in the CHANNEL_BRIDGE event. > > > Thanks for the help, > > > Clint > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/781e1fbe/attachment.html From cal.leeming at simplicitymedialtd.co.uk Tue May 28 20:19:08 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Tue, 28 May 2013 17:19:08 +0100 Subject: [Freeswitch-users] Better approach for multi-tenant SIP testing (part 1) In-Reply-To: References: Message-ID: Thank you! Hopefully soon (finding spare time these days is like stumbling across a pile of gold dust!) Cal On Tue, May 28, 2013 at 5:01 PM, Cesar Bermudez wrote: > Nice Work !!! , when the second part come? > Regards > > On Tue, May 28, 2013 at 8:03 AM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming at simplicitymedialtd.co.uk> wrote: > >> Hello all, >> >> After nearly a year of active development on building our own voice >> platform, we've adopted some cool tricks to speed up the workflow and make >> testing less tedious. >> >> Over the next few months I'll be posting more articles on the tricks we >> have used and the lessons we have learnt. >> >> Here is part 1 - a write up about a neat hack we used to virtualize phone >> testing; >> >> http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing >> >> For those that don't have time to read, here is a screenshot; >> http://i.imgur.com/e9kUdxg.jpg >> >> This trick, which we've used in development for approx 6 months, has >> saved countless hours in development overheads, with one-click start/stop >> functionality and all the necessary modules right there in one place. >> >> Of course, this should never be used as a full replacement as nothing >> beats a proper hardware test to catch those weird edge cases, but it does >> allow you to do the majority of your testing (95%) from a single screen >> rather than a desk full of phones. It also makes attaching wireshark a lot >> easier, without having to use span ports and the alike. >> >> Hope this helps someone else! >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/dfcdf4bf/attachment.html From msc at freeswitch.org Tue May 28 20:19:17 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 09:19:17 -0700 Subject: [Freeswitch-users] voicemail is not working In-Reply-To: <1369378771206-7591043.post@n2.nabble.com> References: <1369378771206-7591043.post@n2.nabble.com> Message-ID: The best thing here would be to look at a debug log of the call. In almost all cases the logs will tell you what is happening right before the call is disconnected. If you want to share your logs with us then use pastebin.freeswitch.org. Put the log lines into a new pastebin and select "FreeSWITCH Log" as the highlighting. Include the pastebin URL in a reply to this thread. -MC On Thu, May 23, 2013 at 11:59 PM, juned wrote: > Hi All, > > I am newbie to FS. so as a startup i have installed FS in mu local system > to > test out the basic functionality and features. so i have registered default > users 1000 and 1001 in softphone( twinkle > ). registration was successful > and > calls was also fine but when i tried to check voicemail then it didn't > worked. > > what i did to test it out voicemail is, did call to 1001 and let it rang so > after 30 second if no answer is there then voicemail will be activated but > calls are released after 30 seconds. > > As per documentation i came to know that by default voicemail is activated > in default extensions is it so ? or i am missing something. > > Please point me to right direction, i want to have such a dialplan in which > user can leave voicemail in cases of busy,unavailable and not answering. > > Thanks & Regards > Juned > > > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/voicemail-is-not-working-tp7591043.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://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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/584361fd/attachment.html From msc at freeswitch.org Tue May 28 20:30:41 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 09:30:41 -0700 Subject: [Freeswitch-users] phone hangs when placing internal calls In-Reply-To: <87vc68z9xw.fsf@mushroom.localdomain> References: <87vc68z9xw.fsf@mushroom.localdomain> Message-ID: Hi Chris, Welcome to the wacky world of telecom! It can be painful but when you finally get things working it's pretty awesome. In your case I suspect a FreeSWITCH log of the call from start to finish would be most useful. A lot of info is available on this handy wiki page. I would bookmark that page if I were you because you'll probably use a lot of the tips presented there. Although the page is about "report bugs" it really starts with the debugging process itself which almost always means gathering information, analyzing it, and asking for help. Using the information on that page, see if you can make a test call, get the FreeSWITCH log for that call, and put it on pastebin. The folks here will be happy to lend a hand with your analysis. You might also want to pick up the brand new FreeSWITCH 1.2 book . -MC On Fri, May 24, 2013 at 8:14 AM, Chris Brannon wrote: > Hello, > I'm very new to freeswitch and telephony in general. > I've managed to register phones with my server, but when I place test > calls, the phone hangs, even though fs_cli tells me that the call was > completed successfully. > Also, everything looks fine when I run sip trace. > FWIW, I've tried both linphone on Linux and csipsimple on Android. > All of this is taking place on my LAN, so NAT isn't an issue. > I started looking at traffic with tcpdump, and here's what I found. > The phone authenticates to FreeSwitch and initiates the call. > FreeSwitch sends a "100 trying" SIP message, followed by a UDP packet > containing what appears to be SDP. It has no SIP header. > This is followed by RTP from > FreeSwitch to the phone. There is never a "200 OK" SIP message. > > Next, I ran linphone in debugging mode. I'll include a link to the log, > in the hope that it will be helpful. Apparently, linphone isn't parsing > the packet after the 100 trying message. Any idea what is going on > here? > > http://the-brannons.com/linphone.log > > -- Chris > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/8ba1c7a3/attachment-0001.html From marketing at cluecon.com Tue May 28 20:56:36 2013 From: marketing at cluecon.com (Michael Collins) Date: Tue, 28 May 2013 09:56:36 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Happy Tuesday to you all! It is a short week for us here in the U.S. as yesterday was a national holiday. The first order of business: I am happy to let you know that Packt Publishing has informed us that the new FreeSWITCH 1.2 book is available as of May 24th! Congrats to the FreeSWITCH team for getting yet another book published. This is the third one we've done with Packt. Also, many thanks to all those who spent time answering questions and helping us with the technical reviewing process. With the new book completed we focus our attention on other things, not the least of which is ClueCon 2013 . Starting this week, those who've submitted talk proposals will be receiving the official acceptance of their presentations, including the day and time for the presentation. If you have not already submitted a talk proposal please do so right away as we have a limited number of available speaking slots. Our weekly conference calls have been lively with discussions on various topics suggested by those calling in. On last week's callwe had a particularly nice discussion about mod_skinny . A relatively new community member, Nathan Neulinger, has really done a nice job of picking up the torch for mod_skinny. If you have a need to use Cisco phones with FreeSWITCH using SCCPthen definitely listen to last week's discussion. This week we have Martin from the VoIPMonitor.org project. We look forward to learning more about VoIPMonitor and how it can help us with keeping tabs on our VoIP servers. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/4d007d34/attachment.html From msc at freeswitch.org Tue May 28 21:05:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 10:05:40 -0700 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: <519F5CCD.8000609@mst.edu> <519F6087.3030706@mst.edu> Message-ID: What does your originate string look like? -MC On Sun, May 26, 2013 at 10:49 PM, Ashish gautam wrote: > Hi Nathan, > > Even setting the api_hangup_hook=perl hook.pl in the originate string > does not work. hook.pl does not get executed on hangup. It has to be done > some other way I guess. > > Thanks. > > > On Fri, May 24, 2013 at 6:13 PM, Nathan Neulinger wrote: > >> I don't think that's going to do what you want... (May be wrong.) >> >> I think that continue_on_fail is only going to apply to the rules for the >> received call on this extension, not the received call on the outgoing leg. >> >> i.e. there are no dialplan rules in effect for the outgoing call that you >> initiated, and that's where the failure is occurring. For these dialplan >> rules, I think the only failure would be if your IVR (I assume that's was >> ash.pl is) didn't answer. >> >> Like I said, not certain of this, maybe some else can chime in, but I >> think you're going to have to handle that failure as a part of your >> originate on the outbound call. Something like putting >> >> originate {api_hangup_hook=perl hook.pl}sofia/..... >> >> Where you cause the call to take place. >> >> -- Nathan >> >> >> On 05/24/2013 07:37 AM, Ashish gautam wrote: >> >>> I am generating an outgoing call through mod_event_socket and then >>> transferring it to this dialplan. >>> >>> On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger >> nneul at mst.edu>> wrote: >>> >>> I may be misunderstanding - but where are you causing it to ring a >>> device? >>> >>> You've told it to internally answer the call, and then not do >>> anything. There's no bridging to an actual extension. >>> >>> Only thing I see that would happen is it running perl/ash.pl < >>> http://ash.pl>, unclear if it would in term execute >>> hook.pl when that script finished (I don't know >>> what that behavior is expected to be). >>> >>> >>> -- Nathan >>> >>> >>> On 05/24/2013 07:17 AM, Ashish gautam wrote: >>> >>> Hi, >>> >>> I have a dialplan as follows: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> when the called party does not pick up the phone or is busy, the >>> dialplan does not proceed and hook.pl >>> >>> >>> >>> does not get executed. >>> >>> Please help >>> -- >>> Ashish Gautam >>> >>> IVR Developer >>> >>> Nucleus Microsystems (Pvt.) Ltd. >>> >>> >>> >>> -- >>> ------------------------------**__----------------------------**-- >>> Nathan Neulinger nneul at mst.edu >>> >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >>> >>> >>> >>> -- >>> Ashish Gautam >>> >>> IVR Developer >>> >>> Nucleus Microsystems (Pvt.) Ltd. >>> >>> Ph. 011 47574758 >>> >> >> -- >> ------------------------------**------------------------------ >> Nathan Neulinger nneul at mst.edu >> Missouri S&T Information Technology (573) 612-1412 >> System Administrator - Architect >> > > > > -- > Ashish Gautam > > IVR Developer > > Nucleus Microsystems (Pvt.) Ltd. > > Ph. 011 47574758 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/fed4cc19/attachment.html From msc at freeswitch.org Tue May 28 21:07:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 10:07:31 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <8fee6197-5f94-418c-a298-f0004c469d95@email.android.com> <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> Message-ID: On Tue, May 28, 2013 at 12:07 AM, wrote: > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/800dd5db/attachment.html From acrow at integrafin.co.uk Tue May 28 21:58:27 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 28 May 2013 18:58:27 +0100 Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: References: <1369661283368-7591135.post@n2.nabble.com> Message-ID: <51A4F043.1090706@integrafin.co.uk> You should probably use ESL/httapi etc with mod_callcenter API commands to handle adding tiers and agents, then you can have that called whenever you want. Cheers Alex On 28/05/13 09:37, Regis M wrote: > mod_callcenter reads configuration only once from xml_curl. > You have to do a queue reload each time you change the agent/tier > configuration, or manualy doing the modification with > callcenter_config on cli. > > It's not like dialplan or directory configuration which doing a http > request on each use. > > > > > 2013/5/27 Anatolii > > > Hello all! > > Freeswitch version - 1.2.8, installed in Centos. > > Freeswitch always goes for configuration to our server (on which > CakePHP is situated) > > I have problem with callcenter module in freeswitch. > > In mod_xml_curl add two bindings: dialplan and configuration. > > < binding name="configuration"> > > < param name="use-dynamic-url" value="yes"/> > > < param name="gateway-url" > value="http://example.comt/freeswitch/configuration/${our_queue_name}${company_id} > " > bindings="configuration"/> > > < /binding> > > So, when I restart (start) freeswitch or reload mod_callcenter - > FS request default configuration for callcenter and it works great > (FS add queues, agents and tiers). > > After that in dialplan for specific number I set application > callcenter with specific name and set 2 global variables, > *our_queue_name* and *company_id*, which send to me with a name of > queue and company id. -- this works fine > > When I receive from freeswitch request - I configure xml from db > data and send FS callcenter configuration. -- works fine. *FS get > correct xml, but add only queue settings and absolutely ignoring > adding agents and tiers.* -- This is my problem for now. > > I try to solve this situation by using lua scripts - no results. > > > ------------------------------------------------------------------------ > View this message in context: Problem with callcenter > configuration (don't register agents and tiers) > > 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://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 > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130528/b55a36c4/attachment-0001.html From acrow at integrafin.co.uk Tue May 28 22:03:23 2013 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 28 May 2013 19:03:23 +0100 Subject: [Freeswitch-users] Debian build scripts - bootstrap.sh disables mod_siren? Message-ID: <51A4F16B.10508@integrafin.co.uk> Hi all, Just a note that in debian/ubuntu builds mod_siren has been working for quite a while (on the order of several months) for me in both squeeze and testing. Is there any reason why this is disabled? If not I will file a Jira to have it readded. Cheers Alex From krice at freeswitch.org Tue May 28 22:13:24 2013 From: krice at freeswitch.org (Ken Rice) Date: Tue, 28 May 2013 13:13:24 -0500 Subject: [Freeswitch-users] Debian build scripts - bootstrap.sh disables mod_siren? In-Reply-To: <51A4F16B.10508@integrafin.co.uk> Message-ID: This is probably due to the fact that mod_siren doesn't meet debians "Free" definition, I would personally think it should be enabled by default On 5/28/13 1:03 PM, "Alex Crow" wrote: > Hi all, > > Just a note that in debian/ubuntu builds mod_siren has been working for > quite a while (on the order of several months) for me in both squeeze > and testing. Is there any reason why this is disabled? If not I will > file a Jira to have it readded. > > Cheers > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From ehermouet at bluetel.fr Tue May 28 22:29:35 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 20:29:35 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <8fee 6197-5f94-418c-a298-f0 004c469d95@email.android.com> <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> Message-ID: <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/ab649ec0/attachment.html From msc at freeswitch.org Tue May 28 22:40:16 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 11:40:16 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> Message-ID: I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > ** ** > > show my pastebin**** > > ** ** > > tks mike**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > ** ** > > ** ** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > ** ** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/3034b76c/attachment.html From ehermouet at bluetel.fr Tue May 28 22:48:50 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 20:48:50 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> Message-ID: <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/5626eb5a/attachment-0001.html From dgarcia at anew.com.ve Tue May 28 23:13:57 2013 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 28 May 2013 14:43:57 -0430 Subject: [Freeswitch-users] sip_ph_X- header doesn't work for freetdm-sofia call In-Reply-To: <201305280045030372439@163.com> References: <201305280045030372439@163.com> Message-ID: <51A501F5.7020007@anew.com.ve> Hi Xiao I not quite sure what are you trying to accomplish. Where are you setting the "sip_ph_X" header? at dialplan level? script (lua, js, etc)? Why you dont "atomize" your actions? For example, at dialplan level: 1. First grab what freetdm give you. Use info app 2. Then capture the channel variable into a new variable 3. When you bridge/transfer the call to the sip side, make the channel setting 4. Use again info app to validate 5. Also check, where are you setting the "sip_ph_X" header, perhaps you are setting it in the wrong leg On 5/27/2013 12:15 PM, xiaofengcanyuexp at 163.com wrote: > Dear support > > I got a problem to pass through SS7 info in SIP messages. Here is the code change and test call flow: > -------------------- > Code change and log: > -------------------- > 1. At sangoma module:intercept ACM in ftmod_sangoma_ss7_*.c, parse the BCI and add it to the ftdmchan->call_data: > sngss7_add_var(sngss7_info, "ss7_acm_bci_hex", val); > > 2. At freetdm module mod_freetdm.c, get the variable and set it to channel. Of which, the "prefix" is 'sip_ph_X-': > var_value = ftdm_sigmsg_get_var(sigmsg, "ss7_acm_bci_hex"); > if (!ftdm_strlen_zero(var_value)) > { > snprintf(variable_name, "%sTEST-BCI-HEX", prefix); > switch_channel_set_variable_printf(channel, variable_name, "%s", var_value); > } > > >From the test call log: > > 2013-05-27 07:03:47.747361 [INFO] ftmod_sangoma_ss7_handle.c:362 [s1c1][1:1] [CIC:1]Rx ACM > 2013-05-27 07:03:47.747361 [DEBUG] ftmod_sangoma_ss7_support.c:2920 [s1c1][1:1] ACM BCI parameter Hex: 0x1636 > 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5513 Get ACM BCI parameter from sangoma > 2013-05-27 09:00:47.127357 [DEBUG] mod_freetdm.c:5530 sip_ph_X-TEST-BCI-HEX:1636 > --------------------- > My purpose: > --------------------- > What I want to do is the encapsulate the sip_ph_X-TEST-BCI-HEX in the 183 SIP message. Like handling IAM in ftdm_channel_from_event(), > once it sets the the variable like below, the "X-FreeTDM-SpanNumber" will automatically attached in INVITE message. > switch_channel_set_variable_printf(channel, "sip_h_X-FreeTDM-SpanNumber", "%d", spanid); > > --------------------- > Problems: > --------------------- > While, what's strange are: > 1. I add one line in dialplan/default.xml to log the variable with: > > The freeswitch.log show ${sip_ph_X-TRUSTID-BCI-HEX} is empty > 2. The 183 message does not attach the 'X-TRUSTID-BCI-HEX'. > > Can someone help to figure out what's wrong with it? > > Thanks > ------------------- > 2013-05-28 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2242 / Virus Database: 3184/5853 - Release Date: 05/24/13 > > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/3be97289/attachment.html From msc at freeswitch.org Tue May 28 23:27:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 12:27:51 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <75afbd9d0cb84712d826d6d3fdf2102c@bluetel.fr> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> Message-ID: Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: > Ok i remove it**** > > ** ** > > But the samedi**** > > http://pastebin.freeswitch.org/20981**** > > ** ** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 20:40 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > I think this might not be helpful: > > EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf()**** > > That's for doing inband DTMF. I would remove that altogether.**** > > -MC**** > > ** ** > > On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet > wrote:**** > > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > **** > > show my pastebin**** > > **** > > tks mike**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > **** > > **** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > **** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/098d4013/attachment.html From ehermouet at bluetel.fr Tue May 28 23:35:39 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 21:35:39 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <75af bd9d0cb84712d826d6d3fd f2102c@bluetel.fr> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> Message-ID: <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/3c8e3743/attachment-0001.html From msc at freeswitch.org Tue May 28 23:43:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 12:43:12 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> Message-ID: Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: > We have **** > > ** ** > > Phone ---? 3cx serveur connected to voip provider ------? voip provider > is freeswitch server ---? my carrier on freeswitch?.**** > > ** ** > > On the other way it?s work**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:28 > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > Question: what is the topology here? What is the phone and what is the > carrier? And which leg needs to have the INFO DTMFs?**** > > -MC**** > > ** ** > > On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet > wrote:**** > > Ok i remove it**** > > **** > > But the samedi**** > > http://pastebin.freeswitch.org/20981**** > > **** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 20:40 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > I think this might not be helpful: > > EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf()**** > > That's for doing inband DTMF. I would remove that altogether.**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet > wrote:**** > > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > **** > > show my pastebin**** > > **** > > tks mike**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > **** > > **** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > **** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/7f45f67c/attachment.html From ehermouet at bluetel.fr Tue May 28 23:49:53 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 21:49:53 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> Message-ID: <0a8601ce5bdc$86a053f0$93e0fbd0$@bluetel.fr> By th phone on 3cx server. If i try another voip provider, so not fs carrier it?s works. De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/92440ccd/attachment-0001.html From ehermouet at bluetel.fr Tue May 28 23:57:22 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 21:57:22 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> Message-ID: <0a8b01ce5bdd$92143160$b63c9420$@bluetel.fr> Mike i try more simple I connect directly my phone to fs, and it?s the same result. No dtmf detected De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/f47edbe3/attachment.html From ehermouet at bluetel.fr Wed May 29 00:02:24 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 22:02:24 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> Message-ID: <0a9001ce5bde$46553660$d2ffa320$@bluetel.fr> I do another interesting test Here my shema Phone 1--? fs ----- > server 3cx ---? phone 2 Call from phone 1 directly to phone 2 (without use my fs carrier) not work It?s not 3cx prob, no my carrier prob but fs prob but where :?( De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/a5389623/attachment-0001.html From msc at freeswitch.org Wed May 29 00:16:23 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 13:16:23 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <0a9001ce5bde$46553660$d2ffa320$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9001ce5bde$46553660$d2ffa320$@bluetel.fr> Message-ID: You need to confirm how the DTMF is reaching FreeSWITCH. Is it inband? RFC2833? SIP INFO? Do you have a pcap w/ RTP of the call leg from the phone to the FS server? -MC On Tue, May 28, 2013 at 1:02 PM, Erwan Hermouet wrote: > I do another interesting test**** > > ** ** > > Here my shema**** > > ** ** > > Phone 1--? fs ----- > server 3cx ---? phone 2**** > > Call from phone 1 directly to phone 2 (without use my fs carrier) not work > **** > > ** ** > > It?s not 3cx prob, no my carrier prob but fs prob? but where? :?(**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:43 > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > Where is the DTMF being generated?**** > > -MC**** > > ** ** > > On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet > wrote:**** > > We have **** > > **** > > Phone ---? 3cx serveur connected to voip provider ------? voip provider > is freeswitch server ---? my carrier on freeswitch?.**** > > **** > > On the other way it?s work**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:28**** > > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > Question: what is the topology here? What is the phone and what is the > carrier? And which leg needs to have the INFO DTMFs?**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet > wrote:**** > > Ok i remove it**** > > **** > > But the samedi**** > > http://pastebin.freeswitch.org/20981**** > > **** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 20:40 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > I think this might not be helpful: > > EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf()**** > > That's for doing inband DTMF. I would remove that altogether.**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet > wrote:**** > > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > **** > > show my pastebin**** > > **** > > tks mike**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > **** > > **** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > **** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/0e6f906f/attachment-0001.html From ehermouet at bluetel.fr Wed May 29 00:24:29 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 22:24:29 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> Message-ID: <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> with wireshar i don?t found sip.Method == INFO on the capture. De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/33a7c147/attachment.html From ehermouet at bluetel.fr Wed May 29 00:27:50 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 22:27:50 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad 01ce5bd1$4f44bb40$edce 31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9001ce5bde$46553660$d2ffa320$@bluetel.fr> Message-ID: <0aa001ce5be1$d3926900$7ab73b00$@bluetel.fr> My provider suggest inband Internal.xml External Provider confi on fs De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 22:16 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call You need to confirm how the DTMF is reaching FreeSWITCH. Is it inband? RFC2833? SIP INFO? Do you have a pcap w/ RTP of the call leg from the phone to the FS server? -MC On Tue, May 28, 2013 at 1:02 PM, Erwan Hermouet wrote: I do another interesting test Here my shema Phone 1--? fs ----- > server 3cx ---? phone 2 Call from phone 1 directly to phone 2 (without use my fs carrier) not work It?s not 3cx prob, no my carrier prob but fs prob but where :?( De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/4e3d6d39/attachment-0001.html From ehermouet at bluetel.fr Wed May 29 00:30:46 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 22:30:46 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad 01ce5bd1$4f44bb40$edce 31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9001ce5bde$46553660$d2ffa320$@bluetel.fr> Message-ID: <0aa501ce5be2$3cd79160$b686b420$@bluetel.fr> Pr?cisly my provider said ? you must use inband-info It?s that i try, but localy it?s not work too, and i dont see sip info on wireshark. De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 22:16 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call You need to confirm how the DTMF is reaching FreeSWITCH. Is it inband? RFC2833? SIP INFO? Do you have a pcap w/ RTP of the call leg from the phone to the FS server? -MC On Tue, May 28, 2013 at 1:02 PM, Erwan Hermouet wrote: I do another interesting test Here my shema Phone 1--? fs ----- > server 3cx ---? phone 2 Call from phone 1 directly to phone 2 (without use my fs carrier) not work It?s not 3cx prob, no my carrier prob but fs prob but where :?( De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/e2e4aee4/attachment-0001.html From msc at freeswitch.org Wed May 29 00:42:31 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 13:42:31 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad01ce5bd1$4f44bb40$edce31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> Message-ID: But do you have a tcpdump of an actual call from the phone to FreeSWITCH where the user pressed a DTMF? If so, please put it where we can download it and look at it. If not, please get one and put it where we can download it. -MC On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet wrote: > with wireshar i don?t found sip.Method == INFO on the capture.**** > > ** ** > > ** ** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:43 > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > Where is the DTMF being generated?**** > > -MC**** > > ** ** > > On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet > wrote:**** > > We have **** > > **** > > Phone ---? 3cx serveur connected to voip provider ------? voip provider > is freeswitch server ---? my carrier on freeswitch?.**** > > **** > > On the other way it?s work**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:28**** > > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > Question: what is the topology here? What is the phone and what is the > carrier? And which leg needs to have the INFO DTMFs?**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet > wrote:**** > > Ok i remove it**** > > **** > > But the samedi**** > > http://pastebin.freeswitch.org/20981**** > > **** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 20:40 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > I think this might not be helpful: > > EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf()**** > > That's for doing inband DTMF. I would remove that altogether.**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet > wrote:**** > > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > **** > > show my pastebin**** > > **** > > tks mike**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > **** > > **** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > **** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/118c505a/attachment.html From tculjaga at gmail.com Wed May 29 00:50:52 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 28 May 2013 22:50:52 +0200 Subject: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE Message-ID: hello im wondering what should we track as hangup event ... CHANNEL_HANGUP or CHANNEL_HANGUP_COMPLETE ... i have a situation where A calls FS, an ESL application answers this call, originates a new call to B and bridges A and B... than ESL originate another call towards C and joins all 3 channels into a conference. now , A hangs up, FS sends just CHANNEL_HANGUP event ... not CHANNEL_HANGUP_COMPLETE ... im wondering if it is supposed to be like that or we need to get CHANNEL_HANGUP_COMPLETE ? T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/ed50aefa/attachment-0001.html From ehermouet at bluetel.fr Wed May 29 00:52:11 2013 From: ehermouet at bluetel.fr (Erwan Hermouet) Date: Tue, 28 May 2013 22:52:11 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <09ad 01ce5bd1$4f44bb40$edce 31c0$@bluetel.fr> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> Message-ID: <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Yes http://Bluetelconnect.fr/tcpdumpdtmf.log De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 22:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call But do you have a tcpdump of an actual call from the phone to FreeSWITCH where the user pressed a DTMF? If so, please put it where we can download it and look at it. If not, please get one and put it where we can download it. -MC On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet wrote: with wireshar i don?t found sip.Method == INFO on the capture. De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:43 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Where is the DTMF being generated? -MC On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet wrote: We have Phone ---? 3cx serveur connected to voip provider ------? voip provider is freeswitch server ---? my carrier on freeswitch . On the other way it?s work De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 21:28 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call Question: what is the topology here? What is the phone and what is the carrier? And which leg needs to have the INFO DTMFs? -MC On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet wrote: Ok i remove it But the samedi http://pastebin.freeswitch.org/20981 De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 20:40 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call I think this might not be helpful: EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf() That's for doing inband DTMF. I would remove that altogether. -MC On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet wrote: Now i don?t see dtmf ????????? :?( http://pastebin.freeswitch.org/20980 show my pastebin tks mike De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Michael Collins Envoy? : mardi 28 mai 2013 19:08 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] DTMF outbound call On Tue, May 28, 2013 at 12:07 AM, wrote: Tks Mike we are try with info on log i see that 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet for [1] ts=141083 dur=160/160/2240 seq=62753 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF 1:2240 but it's not working again :'( tks Call logs? -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/d6a06d6a/attachment-0001.html From msc at freeswitch.org Wed May 29 01:28:01 2013 From: msc at freeswitch.org (Michael Collins) Date: Tue, 28 May 2013 14:28:01 -0700 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Message-ID: I see the RTP streams but there's no audio in them. In other words, there's "empty sound" as if both sides are muted. Make another test call and say something. Also, press several digits on the 3CX phone after the call is established. -MC On Tue, May 28, 2013 at 1:52 PM, Erwan Hermouet wrote: > Yes**** > > ** ** > > http://Bluetelconnect.fr/tcpdumpdtmf.log**** > > ** ** > > ** ** > > ** ** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 22:43 > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > ** ** > > But do you have a tcpdump of an actual call from the phone to FreeSWITCH > where the user pressed a DTMF? If so, please put it where we can download > it and look at it. If not, please get one and put it where we can download > it.**** > > -MC**** > > ** ** > > On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet > wrote:**** > > with wireshar i don?t found sip.Method == INFO on the capture.**** > > **** > > **** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:43**** > > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > Where is the DTMF being generated?**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet > wrote:**** > > We have **** > > **** > > Phone ---? 3cx serveur connected to voip provider ------? voip provider > is freeswitch server ---? my carrier on freeswitch?.**** > > **** > > On the other way it?s work**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 21:28**** > > > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > Question: what is the topology here? What is the phone and what is the > carrier? And which leg needs to have the INFO DTMFs?**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet > wrote:**** > > Ok i remove it**** > > **** > > But the samedi**** > > http://pastebin.freeswitch.org/20981**** > > **** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 20:40 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > I think this might not be helpful: > > EXECUTE sofia/internal/12345 at bluetelconnect.fr:5060 start_dtmf()**** > > That's for doing inband DTMF. I would remove that altogether.**** > > -MC**** > > **** > > On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet > wrote:**** > > Now i don?t see dtmf ????????? :?(**** > > http://pastebin.freeswitch.org/20980**** > > **** > > show my pastebin**** > > **** > > tks mike**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Michael > Collins > *Envoy? :* mardi 28 mai 2013 19:08 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] DTMF outbound call**** > > **** > > **** > > **** > > On Tue, May 28, 2013 at 12:07 AM, wrote:**** > > Tks Mike > > we are try with info > > on log i see that > 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start packet > for [1] ts=141083 dur=160/160/2240 seq=62753 > 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF > 1:2240 > > but it's not working again :'( > > tks > > **** > > **** > > Call logs?**** > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130528/cc0fb4f2/attachment-0001.html From yehavi.bourvine at gmail.com Wed May 29 08:09:44 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 29 May 2013 07:09:44 +0300 Subject: [Freeswitch-users] FreeSWITCH 1.0.X.... Really people? In-Reply-To: References: Message-ID: Better late than never... I am glad to say that two weeks ago I moved from 1.0.x to 1.2.9 (and then to GIT head) and indeed presence works ok now. Thanks for all the good work! Regards, __Yehavi: 2013/3/28 Ken Rice > There has been huge amounts of efforts dedicated to presence in the last > 6 months or so... (I would say Anthony and team spent a good 1000 man hours > working on it (if not more) and its better then ever now... > > Also as another commenter said, Jira numbers? > > > > On 3/28/13 11:44 AM, "Yehavi Bourvine" wrote: > > I'll tell you why I am stuck with 1.0.x: Presence is somewhat erratic in > newer versions. In my lab system it works perfectly ok. When I put it on > production, I have to revert at the end of the day to the old version, and > then I cannot try it again for at least half a year (until my users get > calm down and forget the last time...). > I plan next month to try again with the latest GIT that will be available > at that time... > > Regards, __Yehavi: > > 2013/3/28 Ken Rice > > So I have noticed lately that tons of people are still trying to install > and run FreeSWITCH 1.0.6 and other 1.0.x versions (like 1.0.7). This is a > *VERY* bad idea... > * > *These old versions of FreeSWITCH have known vulnerabilities in them and > are no longer supported by FreeSWITCH.org period. The 1.2 branch has been > released for quite a while now, and is making excellent progress in the > terms of both stability and reliability. > > So now, if you post a Jira for help on FreeSWITCH 1.0 the response you > will get a closed bug with the comments to update and try again. > If you post to the mailing list you will get told to update and try again > If you ask on irc you?ll get told to update and try again (after getting > laughed at and made fun of) > > K > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/a4b20459/attachment.html From yehavi.bourvine at gmail.com Wed May 29 08:14:52 2013 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 29 May 2013 07:14:52 +0300 Subject: [Freeswitch-users] Freeswitch forks new processes? Message-ID: Hello, I have a problem with 1.2.9 (still presented in GIT head from last week); under 40-50 sessions load, incoming sessions starts getting hung in RINGING state. I did not open a JIRA yet as it does not happens when DEBUG level is active, but I noticed a strange behaviour: when it happens I see more than one freeswitch process. During normal work I have only one. When does freeswitch creates more than one process? BTW, I run it with -nc -nonat Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/76db1e6d/attachment.html From ashish at nms.co.in Wed May 29 08:58:19 2013 From: ashish at nms.co.in (Ashish gautam) Date: Wed, 29 May 2013 10:28:19 +0530 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: <519F5CCD.8000609@mst.edu> <519F6087.3030706@mst.edu> Message-ID: The originate string is like this: api originate {voiceMessageID=$voiceMsgID,respreqd=$response_required,mobnum=$mobilenum,lang=$language,ignore_early_media=true,continue_on_fail=true}freetdm/1/a/$mobilenum 47673501 XML public On Tue, May 28, 2013 at 10:35 PM, Michael Collins wrote: > What does your originate string look like? > -MC > > > On Sun, May 26, 2013 at 10:49 PM, Ashish gautam wrote: > >> Hi Nathan, >> >> Even setting the api_hangup_hook=perl hook.pl in the originate string >> does not work. hook.pl does not get executed on hangup. It has to be >> done some other way I guess. >> >> Thanks. >> >> >> On Fri, May 24, 2013 at 6:13 PM, Nathan Neulinger wrote: >> >>> I don't think that's going to do what you want... (May be wrong.) >>> >>> I think that continue_on_fail is only going to apply to the rules for >>> the received call on this extension, not the received call on the outgoing >>> leg. >>> >>> i.e. there are no dialplan rules in effect for the outgoing call that >>> you initiated, and that's where the failure is occurring. For these >>> dialplan rules, I think the only failure would be if your IVR (I assume >>> that's was ash.pl is) didn't answer. >>> >>> Like I said, not certain of this, maybe some else can chime in, but I >>> think you're going to have to handle that failure as a part of your >>> originate on the outbound call. Something like putting >>> >>> originate {api_hangup_hook=perl hook.pl}sofia/..... >>> >>> Where you cause the call to take place. >>> >>> -- Nathan >>> >>> >>> On 05/24/2013 07:37 AM, Ashish gautam wrote: >>> >>>> I am generating an outgoing call through mod_event_socket and then >>>> transferring it to this dialplan. >>>> >>>> On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger >>> nneul at mst.edu>> wrote: >>>> >>>> I may be misunderstanding - but where are you causing it to ring a >>>> device? >>>> >>>> You've told it to internally answer the call, and then not do >>>> anything. There's no bridging to an actual extension. >>>> >>>> Only thing I see that would happen is it running perl/ash.pl < >>>> http://ash.pl>, unclear if it would in term execute >>>> hook.pl when that script finished (I don't know >>>> what that behavior is expected to be). >>>> >>>> >>>> -- Nathan >>>> >>>> >>>> On 05/24/2013 07:17 AM, Ashish gautam wrote: >>>> >>>> Hi, >>>> >>>> I have a dialplan as follows: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> when the called party does not pick up the phone or is busy, >>>> the dialplan does not proceed and hook.pl >>>> >>>> >>>> >>>> does not get executed. >>>> >>>> Please help >>>> -- >>>> Ashish Gautam >>>> >>>> IVR Developer >>>> >>>> Nucleus Microsystems (Pvt.) Ltd. >>>> >>>> >>>> >>>> -- >>>> ------------------------------**__----------------------------**-- >>>> Nathan Neulinger nneul at mst.edu >>>> >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>>> >>>> >>>> >>>> -- >>>> Ashish Gautam >>>> >>>> IVR Developer >>>> >>>> Nucleus Microsystems (Pvt.) Ltd. >>>> >>>> Ph. 011 47574758 >>>> >>> >>> -- >>> ------------------------------**------------------------------ >>> Nathan Neulinger nneul at mst.edu >>> Missouri S&T Information Technology (573) 612-1412 >>> System Administrator - Architect >>> >> >> >> >> -- >> Ashish Gautam >> >> IVR Developer >> >> Nucleus Microsystems (Pvt.) Ltd. >> >> Ph. 011 47574758 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Ashish Gautam IVR Developer Nucleus Microsystems (Pvt.) Ltd. Ph. 011 47574758 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/b27a47ad/attachment-0001.html From alex at opensystems.net.au Wed May 29 09:59:42 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Wed, 29 May 2013 13:59:42 +0800 Subject: [Freeswitch-users] Gateway Call Limits Message-ID: Hi, I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. What's the best way to handle it for an outgoing only system. I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/6c2d8e4a/attachment.html From alex at opensystems.net.au Wed May 29 10:07:36 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Wed, 29 May 2013 14:07:36 +0800 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: Message-ID: I've implemented this in default.xml hoping to limit each of my two gateways to 150. Based on what's in http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so hopefully that works. *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 13:59, Alex Ynema wrote: > Hi, > I'm trying to figure out how to limit the number of calls a Gateway is > allowed to use. Our Sip provider has provided up with 200 which I need to > set within the system somehow. > What's the best way to handle it for an outgoing only system. > I've been trying to figure out how how to configure 'Rate limiting > concurrent outgoing calls via a provider' which is mentioned in the wiki > a bit but nothing specific on what to actually do. > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/2cf16c4f/attachment.html From krice at freeswitch.org Wed May 29 10:12:07 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 29 May 2013 01:12:07 -0500 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: Message-ID: Limit ... See the wiki On 5/29/13 12:59 AM, "Alex Ynema" wrote: > Hi, > I'm trying to figure out how to limit the number of calls a Gateway is allowed > to use. Our Sip provider has provided up with 200 which I need to set within > the system somehow. > What's the best way to handle it for an outgoing only system. > I've been trying to figure out how how to configure 'Rate limiting concurrent > outgoing calls via a provider' which is mentioned in the wiki a bit but > nothing specific on what to actually do. > > Alex Ynema?|?IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/7f05012f/attachment.html From avi at avimarcus.net Wed May 29 10:13:36 2013 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 29 May 2013 09:13:36 +0300 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: Message-ID: ... just note that's stored in a database (db) not ram (hash) so if you don't need to share it / have persistence, just store it in ram. -Avi On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: > I've implemented this in default.xml hoping to limit each of my two > gateways to 150. > Based on what's in > http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so > hopefully that works. > > > > > data="sofia/gateway/zetta-cisco-1/${destnum}" /> > > > > > > data="sofia/gateway/zetta-cisco-2/${destnum}" /> > > > > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 13:59, Alex Ynema wrote: > >> Hi, >> I'm trying to figure out how to limit the number of calls a Gateway is >> allowed to use. Our Sip provider has provided up with 200 which I need to >> set within the system somehow. >> What's the best way to handle it for an outgoing only system. >> I've been trying to figure out how how to configure 'Rate limiting >> concurrent outgoing calls via a provider' which is mentioned in the wiki >> a bit but nothing specific on what to actually do. >> >> *Alex Ynema** *| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/58e6dc50/attachment-0001.html From alex at opensystems.net.au Wed May 29 10:27:48 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Wed, 29 May 2013 14:27:48 +0800 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: Message-ID: Cheers Avi I've now changed that to hash as I don't need it to be persistent. What should I see in the clie to confirm this is working without attempting 150+ calls Basically I've added this to my default.xml *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 14:13, Avi Marcus wrote: > ... just note that's stored in a database (db) not ram (hash) so if you > don't need to share it / have persistence, just store it in ram. > > -Avi > > On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: > >> I've implemented this in default.xml hoping to limit each of my two >> gateways to 150. >> Based on what's in >> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >> hopefully that works. >> >> >> >> >> > data="sofia/gateway/zetta-cisco-1/${destnum}" /> >> >> >> >> >> >> > data="sofia/gateway/zetta-cisco-2/${destnum}" /> >> >> >> >> >> *Alex Ynema** *| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 13:59, Alex Ynema wrote: >> >>> Hi, >>> I'm trying to figure out how to limit the number of calls a Gateway is >>> allowed to use. Our Sip provider has provided up with 200 which I need to >>> set within the system somehow. >>> What's the best way to handle it for an outgoing only system. >>> I've been trying to figure out how how to configure 'Rate limiting >>> concurrent outgoing calls via a provider' which is mentioned in the >>> wiki a bit but nothing specific on what to actually do. >>> >>> *Alex Ynema** *| IT Consultant >>> alex at opensystems.net.au >>> >>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>> Office: +61 8 9427 2500 >>> Mobile: +61 404 796 894 >>> >>> IT Consultant for Open Systems Support >>> www.opensystems.net.au >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/83f643d2/attachment.html From nbhatti at gmail.com Wed May 29 10:52:29 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 29 May 2013 10:52:29 +0400 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: Message-ID: <51A5A5AD.1010605@gmail.com> Sorry for the thread hijack, but on the other hand, is it possible to limit the number of outgoing CPS? Don't seem to see that either in the wiki or a way to make it work. -- Thanks, Muhammad Naseer Bhatti Alex Ynema wrote: > Cheers Avi I've now changed that to hash as I don't need it to be > persistent. > What should I see in the clie to confirm this is working without > attempting 150+ calls > > Basically I've added this to my default.xml > > > data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> > > > > > data="sofia/gateway/zetta-cisco-1/${destnum}" /> > > > > > > data="sofia/gateway/zetta-cisco-2/${destnum}" /> > > > > > *Alex Ynema***| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 14:13, Avi Marcus > wrote: > > ... just note that's stored in a database (db) not ram (hash) so > if you don't need to share it / have persistence, just store it in > ram. > > -Avi > > On Wed, May 29, 2013 at 9:07 AM, Alex Ynema > > wrote: > > I've implemented this in default.xml hoping to limit each of > my two gateways to 150. > Based on what's in > http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so > hopefully that works. > > > > > data="sofia/gateway/zetta-cisco-1/${destnum}" /> > > > > > > data="sofia/gateway/zetta-cisco-2/${destnum}" /> > > > > > *Alex Ynema***| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 13:59, Alex Ynema > wrote: > > Hi, > I'm trying to figure out how to limit the number of calls > a Gateway is allowed to use. Our Sip provider has provided > up with 200 which I need to set within the system somehow. > What's the best way to handle it for an outgoing only system. > I've been trying to figure out how how to configure 'Rate > limiting concurrent outgoing calls via a provider' which > is mentioned in the wiki a bit but nothing specific on > what to actually do. > > *Alex Ynema***| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/87cd2f2f/attachment-0001.html From jaybinks at gmail.com Wed May 29 11:01:56 2013 From: jaybinks at gmail.com (jay binks) Date: Wed, 29 May 2013 17:01:56 +1000 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: <51A5A5AD.1010605@gmail.com> References: <51A5A5AD.1010605@gmail.com> Message-ID: http://wiki.freeswitch.org/wiki/Limit On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: > > Sorry for the thread hijack, but on the other hand, is it possible to > limit the number of outgoing CPS? Don't seem to see that either in the wiki > or a way to make it work. > > -- > Thanks, > Muhammad Naseer Bhatti > > > > Alex Ynema wrote: > > Cheers Avi I've now changed that to hash as I don't need it to be > persistent. > What should I see in the clie to confirm this is working without > attempting 150+ calls > > Basically I've added this to my default.xml > > > data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> > > > > > data="sofia/gateway/zetta-cisco-1/${destnum}" /> > > > > > > data="sofia/gateway/zetta-cisco-2/${destnum}" /> > > > > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 14:13, Avi Marcus wrote: > >> ... just note that's stored in a database (db) not ram (hash) so if you >> don't need to share it / have persistence, just store it in ram. >> >> -Avi >> >> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >> >>> I've implemented this in default.xml hoping to limit each of my two >>> gateways to 150. >>> Based on what's in >>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>> hopefully that works. >>> >>> >>> >>> >>> >> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>> >>> >>> >>> >>> >>> >> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>> >>> >>> >>> >>> *Alex Ynema** *| IT Consultant >>> alex at opensystems.net.au >>> >>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>> Office: +61 8 9427 2500 >>> Mobile: +61 404 796 894 >>> >>> IT Consultant for Open Systems Support >>> www.opensystems.net.au >>> >>> >>> On 29 May 2013 13:59, Alex Ynema wrote: >>> >>>> Hi, >>>> I'm trying to figure out how to limit the number of calls a Gateway is >>>> allowed to use. Our Sip provider has provided up with 200 which I need to >>>> set within the system somehow. >>>> What's the best way to handle it for an outgoing only system. >>>> I've been trying to figure out how how to configure 'Rate limiting >>>> concurrent outgoing calls via a provider' which is mentioned in the >>>> wiki a bit but nothing specific on what to actually do. >>>> >>>> *Alex Ynema** *| IT Consultant >>>> alex at opensystems.net.au >>>> >>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>> Office: +61 8 9427 2500 >>>> Mobile: +61 404 796 894 >>>> >>>> IT Consultant for Open Systems Support >>>> www.opensystems.net.au >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 > > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/941056d0/attachment.html From ehermouet at bluetel.fr Wed May 29 11:04:28 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Wed, 29 May 2013 09:04:28 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Message-ID: Michael it was firewall problem. now i have the audio, but i don't see dtmf again. on wireshark i don't see too. http://bluetelconnect.fr/tcpdumpdtmf3.log tks Le 2013-05-28 23:28, Michael Collins a ?crit?: > I see the RTP streams but theres no audio in them. In other words, > theres "empty sound" as if both sides are muted. Make another test > call and say something. Also, press several digits on the 3CX phone > after the call is established. > > -MC > > On Tue, May 28, 2013 at 1:52 PM, Erwan Hermouet [85]> wrote: > >> Yes >> >> ? >> >> http://Bluetelconnect.fr/tcpdumpdtmf.log [1] >> >> ? >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [2] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [3]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 22:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> But do you have a tcpdump of an actual call from the phone to >> FreeSWITCH where the user pressed a DTMF? If so, please put it where >> we can download it and look at it. If not, please get one and put it >> where we can download it. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet >> wrote: >> >> with wireshar i don?t found sip.Method == INFO on the capture. >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [5] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [6]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Where is the DTMF being generated? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet >> wrote: >> >> We have >> >> ? >> >> Phone ---? 3cx serveur connected to voip provider ?------? voip >> provider is freeswitch server ---? my carrier on freeswitch?. >> >> ? >> >> On the other way it?s work >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [8] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [9]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:28 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Question: what is the topology here? What is the phone and what is >> the carrier? And which leg needs to have the INFO DTMFs? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet >> wrote: >> >> Ok i remove it >> >> ? >> >> But the samedi >> >> http://pastebin.freeswitch.org/20981 [11] >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [12] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [13]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 20:40 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> I think this might not be helpful: >> >> EXECUTE sofia/internal/12345 at bluetelconnect.fr [14]:5060 >> start_dtmf() >> >> Thats for doing inband DTMF. I would remove that altogether. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet >> wrote: >> >> Now i don?t see dtmf???????????:?( >> >> http://pastebin.freeswitch.org/20980 [16] >> >> ? >> >> show my pastebin >> >> ? >> >> tks mike >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [17] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [18]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 19:08 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> ? >> >> ? >> >> On Tue, May 28, 2013 at 12:07 AM, wrote: >> >> >> Tks Mike >> >> we are try with info >> >> on log i see that >> 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start >> packet >> for [1] ts=141083 dur=160/160/2240 seq=62753 >> 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF >> 1:2240 >> >> but its not working again :( >> >> tks >> >> ? >> >> Call logs? >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [20] >> http://www.ClueCon.com [21] >> http://www.OSTAG.org [22] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [23] >> http://www.freeswitchsolutions.com [24] >> >> >> [25] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [26] >> http://wiki.freeswitch.org [27] >> http://www.cluecon.com [28] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [29] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [30] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [31] >> http://www.freeswitch.org [32] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [33] >> http://www.ClueCon.com [34] >> http://www.OSTAG.org [35] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [36] >> http://www.freeswitchsolutions.com [37] >> >> >> [38] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [39] >> http://wiki.freeswitch.org [40] >> http://www.cluecon.com [41] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [42] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [43] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [44] >> http://www.freeswitch.org [45] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [46] >> http://www.ClueCon.com [47] >> http://www.OSTAG.org [48] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [49] >> http://www.freeswitchsolutions.com [50] >> >> >> [51] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [52] >> http://wiki.freeswitch.org [53] >> http://www.cluecon.com [54] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [55] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [56] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [57] >> http://www.freeswitch.org [58] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [59] >> http://www.ClueCon.com [60] >> http://www.OSTAG.org [61] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [62] >> http://www.freeswitchsolutions.com [63] >> >> >> [64] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [65] >> http://wiki.freeswitch.org [66] >> http://www.cluecon.com [67] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [68] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [70] >> http://www.freeswitch.org [71] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [72] >> http://www.ClueCon.com [73] >> http://www.OSTAG.org [74] >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [75] >> http://www.freeswitchsolutions.com [76] >> >> >> [77] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [78] >> http://wiki.freeswitch.org [79] >> http://www.cluecon.com [80] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [81] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [82] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [83] >> http://www.freeswitch.org [84] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [86] > http://www.ClueCon.com [87] > http://www.OSTAG.org [88] > > > > Links: > ------ > [1] http://Bluetelconnect.fr/tcpdumpdtmf.log > [2] mailto:freeswitch-users-bounces at lists.freeswitch.org > [3] mailto:freeswitch-users-bounces at lists.freeswitch.org > [4] mailto:ehermouet at bluetel.fr > [5] mailto:freeswitch-users-bounces at lists.freeswitch.org > [6] mailto:freeswitch-users-bounces at lists.freeswitch.org > [7] mailto:ehermouet at bluetel.fr > [8] mailto:freeswitch-users-bounces at lists.freeswitch.org > [9] mailto:freeswitch-users-bounces at lists.freeswitch.org > [10] mailto:ehermouet at bluetel.fr > [11] http://pastebin.freeswitch.org/20981 > [12] mailto:freeswitch-users-bounces at lists.freeswitch.org > [13] mailto:freeswitch-users-bounces at lists.freeswitch.org > [14] http://bluetelconnect.fr > [15] mailto:ehermouet at bluetel.fr > [16] http://pastebin.freeswitch.org/20980 > [17] mailto:freeswitch-users-bounces at lists.freeswitch.org > [18] mailto:freeswitch-users-bounces at lists.freeswitch.org > [19] mailto:ehermouet at bluetel.fr > [20] http://www.FreeSWITCH.org > [21] http://www.ClueCon.com > [22] http://www.OSTAG.org > [23] mailto:consulting at freeswitch.org > [24] http://www.freeswitchsolutions.com > [25] > [26] http://www.freeswitch.org > [27] http://wiki.freeswitch.org > [28] http://www.cluecon.com > [29] mailto:FreeSWITCH-users at lists.freeswitch.org > [30] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [31] http://lists.freeswitch.org/mailman/options/freeswitch-users > [32] http://www.freeswitch.org > [33] http://www.FreeSWITCH.org > [34] http://www.ClueCon.com > [35] http://www.OSTAG.org > [36] mailto:consulting at freeswitch.org > [37] http://www.freeswitchsolutions.com > [38] > [39] http://www.freeswitch.org > [40] http://wiki.freeswitch.org > [41] http://www.cluecon.com > [42] mailto:FreeSWITCH-users at lists.freeswitch.org > [43] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [44] http://lists.freeswitch.org/mailman/options/freeswitch-users > [45] http://www.freeswitch.org > [46] http://www.FreeSWITCH.org > [47] http://www.ClueCon.com > [48] http://www.OSTAG.org > [49] mailto:consulting at freeswitch.org > [50] http://www.freeswitchsolutions.com > [51] > [52] http://www.freeswitch.org > [53] http://wiki.freeswitch.org > [54] http://www.cluecon.com > [55] mailto:FreeSWITCH-users at lists.freeswitch.org > [56] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [57] http://lists.freeswitch.org/mailman/options/freeswitch-users > [58] http://www.freeswitch.org > [59] http://www.FreeSWITCH.org > [60] http://www.ClueCon.com > [61] http://www.OSTAG.org > [62] mailto:consulting at freeswitch.org > [63] http://www.freeswitchsolutions.com > [64] > [65] http://www.freeswitch.org > [66] http://wiki.freeswitch.org > [67] http://www.cluecon.com > [68] mailto:FreeSWITCH-users at lists.freeswitch.org > [69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [70] http://lists.freeswitch.org/mailman/options/freeswitch-users > [71] http://www.freeswitch.org > [72] http://www.FreeSWITCH.org > [73] http://www.ClueCon.com > [74] http://www.OSTAG.org > [75] mailto:consulting at freeswitch.org > [76] http://www.freeswitchsolutions.com > [77] > [78] http://www.freeswitch.org > [79] http://wiki.freeswitch.org > [80] http://www.cluecon.com > [81] mailto:FreeSWITCH-users at lists.freeswitch.org > [82] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [83] http://lists.freeswitch.org/mailman/options/freeswitch-users > [84] http://www.freeswitch.org > [85] mailto:ehermouet at bluetel.fr > [86] http://www.FreeSWITCH.org > [87] http://www.ClueCon.com > [88] http://www.OSTAG.org From alex at opensystems.net.au Wed May 29 11:17:33 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Wed, 29 May 2013 15:17:33 +0800 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: so setting the limit at 150 was fine but as soon as I set it to 200 I've now hit a problem. Freeswitch has slowly grown to 372 sessions and getting lots of these errors in the cli 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more events..... UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, 606 microseconds FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready 1500 session(s) since startup 372 session(s) - 0 out of max 10 per sec 10000 session(s) max min idle cpu 0.00/100.00 Current Stack Size/Max 240K/8192K *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 15:01, jay binks wrote: > http://wiki.freeswitch.org/wiki/Limit > > > > > On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: > >> >> Sorry for the thread hijack, but on the other hand, is it possible to >> limit the number of outgoing CPS? Don't seem to see that either in the wiki >> or a way to make it work. >> >> -- >> Thanks, >> Muhammad Naseer Bhatti >> >> >> >> Alex Ynema wrote: >> >> Cheers Avi I've now changed that to hash as I don't need it to be >> persistent. >> What should I see in the clie to confirm this is working without >> attempting 150+ calls >> >> Basically I've added this to my default.xml >> >> >> > data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >> >> >> >> >> > data="sofia/gateway/zetta-cisco-1/${destnum}" /> >> >> >> >> >> >> > data="sofia/gateway/zetta-cisco-2/${destnum}" /> >> >> >> >> >> *Alex Ynema** *| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 14:13, Avi Marcus wrote: >> >>> ... just note that's stored in a database (db) not ram (hash) so if you >>> don't need to share it / have persistence, just store it in ram. >>> >>> -Avi >>> >>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >>> >>>> I've implemented this in default.xml hoping to limit each of my two >>>> gateways to 150. >>>> Based on what's in >>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>> hopefully that works. >>>> >>>> >>>> >>> expression="zetta-cisco-1"> >>>> >>>> >>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>> >>>> >>>> >>>> >>> expression="zetta-cisco-2"> >>>> >>>> >>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>> >>>> >>>> >>>> >>>> *Alex Ynema** *| IT Consultant >>>> alex at opensystems.net.au >>>> >>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>> Office: +61 8 9427 2500 >>>> Mobile: +61 404 796 894 >>>> >>>> IT Consultant for Open Systems Support >>>> www.opensystems.net.au >>>> >>>> >>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>> >>>>> Hi, >>>>> I'm trying to figure out how to limit the number of calls a Gateway is >>>>> allowed to use. Our Sip provider has provided up with 200 which I need to >>>>> set within the system somehow. >>>>> What's the best way to handle it for an outgoing only system. >>>>> I've been trying to figure out how how to configure 'Rate limiting >>>>> concurrent outgoing calls via a provider' which is mentioned in the >>>>> wiki a bit but nothing specific on what to actually do. >>>>> >>>>> *Alex Ynema** *| IT Consultant >>>>> alex at opensystems.net.au >>>>> >>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>> Office: +61 8 9427 2500 >>>>> Mobile: +61 404 796 894 >>>>> >>>>> IT Consultant for Open Systems Support >>>>> www.opensystems.net.au >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >> >> > > > -- > Sincerely > > Jay > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/29b7b296/attachment-0001.html From gmaruzz at gmail.com Wed May 29 11:18:29 2013 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 29 May 2013 09:18:29 +0200 Subject: [Freeswitch-users] Best Installation MOD_GSMOPEN In-Reply-To: References: Message-ID: Thanks Nandy! On Tue, May 28, 2013 at 4:34 AM, Nandy Dagondon wrote: > After wresting the problem installing gsmopen on Centos 6.4 x_64 (** > libctb-2.6.so problem), I feel this information should be added in the > Wiki in the pre-requisites > > ldconfig <-- after installing gsm library > ---- Add this --- > Check if the new library is included > # ldconfig -p | grep gsm > > If no entries found, correct this problem by using the ldconfig examples > at http://linux.101hacks.com/unix/ldconfig/ > > Run ldconfig again > ----- End --- > > In my case, I added this rule file in /etc/ld.so.d directory > # local-libs.conf > /usr/local/lib > > > /nandy > > > > On Tue, Feb 26, 2013 at 2:10 AM, Josue Diaz Cruz wrote: > >> ** >> Good Afternoon. >> >> I beg your pardon if my question is really from a newbee. >> >> I am trying to install gsmopen in a system. I did it with CentOS 6 X64, >> Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. >> Cause i allways have problems in one point of the installation. Depending >> the distro could it be one or other. I was thinking that could be nice to >> give a complete requierements (libs) to install before do anything. >> >> If some one can provide me a easy, simple and not troubles, could be nice: >> >> * System and version >> * List of requierements. >> * Better steps to avoid issues with libraries and compile. >> >> What i tried: >> >> Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when >> try to load mod_gsmopen.so) >> Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with >> the ttyUSB port) >> Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with >> the ttyUSB port) >> >> E169 works perfectly. I have a mobile partner with voicecall and i test >> it calling my land line without issues. >> >> >> Thank you very much for your consideration. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/1c75bc41/attachment.html From nbhatti at gmail.com Wed May 29 11:20:13 2013 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 29 May 2013 11:20:13 +0400 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: <51A5AC2D.3040406@gmail.com> Hmm, so simple. I wonder how I overlooked it :) -- Thanks, Muhammad Naseer Bhatti jay binks wrote: > http://wiki.freeswitch.org/wiki/Limit > > > > > On 29 May 2013 16:52, Muhammad Naseer Bhatti > wrote: > > > Sorry for the thread hijack, but on the other hand, is it possible > to limit the number of outgoing CPS? Don't seem to see that either > in the wiki or a way to make it work. > > -- > Thanks, > Muhammad Naseer Bhatti > > > > Alex Ynema wrote: >> Cheers Avi I've now changed that to hash as I don't need it to be >> persistent. >> What should I see in the clie to confirm this is working without >> attempting 150+ calls >> >> Basically I've added this to my default.xml >> >> >> > data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" >> /> >> >> >> >> >> > data="sofia/gateway/zetta-cisco-1/${destnum}" /> >> >> >> >> >> >> > data="sofia/gateway/zetta-cisco-2/${destnum}" /> >> >> >> >> >> *Alex Ynema***| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 14:13, Avi Marcus > > wrote: >> >> ... just note that's stored in a database (db) not ram (hash) >> so if you don't need to share it / have persistence, just >> store it in ram. >> >> -Avi >> >> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema >> > wrote: >> >> I've implemented this in default.xml hoping to limit each >> of my two gateways to 150. >> Based on what's in >> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >> hopefully that works. >> >> >> > expression="zetta-cisco-1"> >> >> > data="sofia/gateway/zetta-cisco-1/${destnum}" /> >> >> >> >> > expression="zetta-cisco-2"> >> >> > data="sofia/gateway/zetta-cisco-2/${destnum}" /> >> >> >> >> >> *Alex Ynema***| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 13:59, Alex Ynema > > wrote: >> >> Hi, >> I'm trying to figure out how to limit the number of >> calls a Gateway is allowed to use. Our Sip provider >> has provided up with 200 which I need to set within >> the system somehow. >> What's the best way to handle it for an outgoing only >> system. >> I've been trying to figure out how how to configure >> 'Rate limiting concurrent outgoing calls via a >> provider' which is mentioned in the wiki a bit but >> nothing specific on what to actually do. >> >> *Alex Ynema***| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Sincerely > > Jay > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/0f44cabb/attachment-0001.html From ehermouet at bluetel.fr Wed May 29 11:32:05 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Wed, 29 May 2013 09:32:05 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Message-ID: <611e9a2a33af1213dcea277be46a6788@bluetel.fr> In my new test you can see that there is audio but i press key on phone and on my phone i hear sound but on wireshark no this sound key.... strange. Le 2013-05-28 23:28, Michael Collins a ?crit?: > I see the RTP streams but theres no audio in them. In other words, > theres "empty sound" as if both sides are muted. Make another test > call and say something. Also, press several digits on the 3CX phone > after the call is established. > > -MC > > On Tue, May 28, 2013 at 1:52 PM, Erwan Hermouet [85]> wrote: > >> Yes >> >> ? >> >> http://Bluetelconnect.fr/tcpdumpdtmf.log [1] >> >> ? >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [2] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [3]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 22:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> But do you have a tcpdump of an actual call from the phone to >> FreeSWITCH where the user pressed a DTMF? If so, please put it where >> we can download it and look at it. If not, please get one and put it >> where we can download it. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet >> wrote: >> >> with wireshar i don?t found sip.Method == INFO on the capture. >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [5] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [6]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Where is the DTMF being generated? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet >> wrote: >> >> We have >> >> ? >> >> Phone ---? 3cx serveur connected to voip provider ?------? voip >> provider is freeswitch server ---? my carrier on freeswitch?. >> >> ? >> >> On the other way it?s work >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [8] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [9]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:28 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Question: what is the topology here? What is the phone and what is >> the carrier? And which leg needs to have the INFO DTMFs? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet >> wrote: >> >> Ok i remove it >> >> ? >> >> But the samedi >> >> http://pastebin.freeswitch.org/20981 [11] >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [12] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [13]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 20:40 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> I think this might not be helpful: >> >> EXECUTE sofia/internal/12345 at bluetelconnect.fr [14]:5060 >> start_dtmf() >> >> Thats for doing inband DTMF. I would remove that altogether. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet >> wrote: >> >> Now i don?t see dtmf???????????:?( >> >> http://pastebin.freeswitch.org/20980 [16] >> >> ? >> >> show my pastebin >> >> ? >> >> tks mike >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [17] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [18]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 19:08 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> ? >> >> ? >> >> On Tue, May 28, 2013 at 12:07 AM, wrote: >> >> >> Tks Mike >> >> we are try with info >> >> on log i see that >> 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start >> packet >> for [1] ts=141083 dur=160/160/2240 seq=62753 >> 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF >> 1:2240 >> >> but its not working again :( >> >> tks >> >> ? >> >> Call logs? >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [20] >> http://www.ClueCon.com [21] >> http://www.OSTAG.org [22] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [23] >> http://www.freeswitchsolutions.com [24] >> >> >> [25] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [26] >> http://wiki.freeswitch.org [27] >> http://www.cluecon.com [28] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [29] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [30] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [31] >> http://www.freeswitch.org [32] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [33] >> http://www.ClueCon.com [34] >> http://www.OSTAG.org [35] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [36] >> http://www.freeswitchsolutions.com [37] >> >> >> [38] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [39] >> http://wiki.freeswitch.org [40] >> http://www.cluecon.com [41] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [42] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [43] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [44] >> http://www.freeswitch.org [45] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [46] >> http://www.ClueCon.com [47] >> http://www.OSTAG.org [48] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [49] >> http://www.freeswitchsolutions.com [50] >> >> >> [51] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [52] >> http://wiki.freeswitch.org [53] >> http://www.cluecon.com [54] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [55] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [56] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [57] >> http://www.freeswitch.org [58] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [59] >> http://www.ClueCon.com [60] >> http://www.OSTAG.org [61] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [62] >> http://www.freeswitchsolutions.com [63] >> >> >> [64] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [65] >> http://wiki.freeswitch.org [66] >> http://www.cluecon.com [67] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [68] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [70] >> http://www.freeswitch.org [71] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [72] >> http://www.ClueCon.com [73] >> http://www.OSTAG.org [74] >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [75] >> http://www.freeswitchsolutions.com [76] >> >> >> [77] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [78] >> http://wiki.freeswitch.org [79] >> http://www.cluecon.com [80] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [81] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [82] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [83] >> http://www.freeswitch.org [84] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [86] > http://www.ClueCon.com [87] > http://www.OSTAG.org [88] > > > > Links: > ------ > [1] http://Bluetelconnect.fr/tcpdumpdtmf.log > [2] mailto:freeswitch-users-bounces at lists.freeswitch.org > [3] mailto:freeswitch-users-bounces at lists.freeswitch.org > [4] mailto:ehermouet at bluetel.fr > [5] mailto:freeswitch-users-bounces at lists.freeswitch.org > [6] mailto:freeswitch-users-bounces at lists.freeswitch.org > [7] mailto:ehermouet at bluetel.fr > [8] mailto:freeswitch-users-bounces at lists.freeswitch.org > [9] mailto:freeswitch-users-bounces at lists.freeswitch.org > [10] mailto:ehermouet at bluetel.fr > [11] http://pastebin.freeswitch.org/20981 > [12] mailto:freeswitch-users-bounces at lists.freeswitch.org > [13] mailto:freeswitch-users-bounces at lists.freeswitch.org > [14] http://bluetelconnect.fr > [15] mailto:ehermouet at bluetel.fr > [16] http://pastebin.freeswitch.org/20980 > [17] mailto:freeswitch-users-bounces at lists.freeswitch.org > [18] mailto:freeswitch-users-bounces at lists.freeswitch.org > [19] mailto:ehermouet at bluetel.fr > [20] http://www.FreeSWITCH.org > [21] http://www.ClueCon.com > [22] http://www.OSTAG.org > [23] mailto:consulting at freeswitch.org > [24] http://www.freeswitchsolutions.com > [25] > [26] http://www.freeswitch.org > [27] http://wiki.freeswitch.org > [28] http://www.cluecon.com > [29] mailto:FreeSWITCH-users at lists.freeswitch.org > [30] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [31] http://lists.freeswitch.org/mailman/options/freeswitch-users > [32] http://www.freeswitch.org > [33] http://www.FreeSWITCH.org > [34] http://www.ClueCon.com > [35] http://www.OSTAG.org > [36] mailto:consulting at freeswitch.org > [37] http://www.freeswitchsolutions.com > [38] > [39] http://www.freeswitch.org > [40] http://wiki.freeswitch.org > [41] http://www.cluecon.com > [42] mailto:FreeSWITCH-users at lists.freeswitch.org > [43] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [44] http://lists.freeswitch.org/mailman/options/freeswitch-users > [45] http://www.freeswitch.org > [46] http://www.FreeSWITCH.org > [47] http://www.ClueCon.com > [48] http://www.OSTAG.org > [49] mailto:consulting at freeswitch.org > [50] http://www.freeswitchsolutions.com > [51] > [52] http://www.freeswitch.org > [53] http://wiki.freeswitch.org > [54] http://www.cluecon.com > [55] mailto:FreeSWITCH-users at lists.freeswitch.org > [56] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [57] http://lists.freeswitch.org/mailman/options/freeswitch-users > [58] http://www.freeswitch.org > [59] http://www.FreeSWITCH.org > [60] http://www.ClueCon.com > [61] http://www.OSTAG.org > [62] mailto:consulting at freeswitch.org > [63] http://www.freeswitchsolutions.com > [64] > [65] http://www.freeswitch.org > [66] http://wiki.freeswitch.org > [67] http://www.cluecon.com > [68] mailto:FreeSWITCH-users at lists.freeswitch.org > [69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [70] http://lists.freeswitch.org/mailman/options/freeswitch-users > [71] http://www.freeswitch.org > [72] http://www.FreeSWITCH.org > [73] http://www.ClueCon.com > [74] http://www.OSTAG.org > [75] mailto:consulting at freeswitch.org > [76] http://www.freeswitchsolutions.com > [77] > [78] http://www.freeswitch.org > [79] http://wiki.freeswitch.org > [80] http://www.cluecon.com > [81] mailto:FreeSWITCH-users at lists.freeswitch.org > [82] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [83] http://lists.freeswitch.org/mailman/options/freeswitch-users > [84] http://www.freeswitch.org > [85] mailto:ehermouet at bluetel.fr > [86] http://www.FreeSWITCH.org > [87] http://www.ClueCon.com > [88] http://www.OSTAG.org From admin at smallunix.net Wed May 29 11:45:46 2013 From: admin at smallunix.net (arag00rn) Date: Wed, 29 May 2013 00:45:46 -0700 (PDT) Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error In-Reply-To: References: Message-ID: <1369813546562-7591226.post@n2.nabble.com> Hello Brian, did you manage to solve this issue ? I have exactly the same problem. FreeSWITCH (Version 1.5.1b git 16690e4 2013-05-09 19:05:09Z) The external SBC is a Sonus. BR, Andrea -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591226.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ehermouet at bluetel.fr Wed May 29 12:36:15 2013 From: ehermouet at bluetel.fr (ehermouet at bluetel.fr) Date: Wed, 29 May 2013 10:36:15 +0200 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Message-ID: <9f815ba38ddba0bc49713c68cb5e4685@bluetel.fr> Mike i found some interesting things. 3cx server use only rfc2833... so my call start with rfc2833 and i see them on 3cx server... Next i don't see on FS. so i decide to return to fs and change to RFC 2833, now i can see on log all dtmf but not work on last phone. I will contact my provider to said that we must have 2833 on outbound... or maybe you have solution to convert 2833 on info on FS for outbound. Le 2013-05-28 23:28, Michael Collins a ?crit?: > I see the RTP streams but theres no audio in them. In other words, > theres "empty sound" as if both sides are muted. Make another test > call and say something. Also, press several digits on the 3CX phone > after the call is established. > > -MC > > On Tue, May 28, 2013 at 1:52 PM, Erwan Hermouet [85]> wrote: > >> Yes >> >> ? >> >> http://Bluetelconnect.fr/tcpdumpdtmf.log [1] >> >> ? >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [2] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [3]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 22:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> But do you have a tcpdump of an actual call from the phone to >> FreeSWITCH where the user pressed a DTMF? If so, please put it where >> we can download it and look at it. If not, please get one and put it >> where we can download it. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 1:24 PM, Erwan Hermouet >> wrote: >> >> with wireshar i don?t found sip.Method == INFO on the capture. >> >> ? >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [5] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [6]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:43 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Where is the DTMF being generated? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 12:35 PM, Erwan Hermouet >> wrote: >> >> We have >> >> ? >> >> Phone ---? 3cx serveur connected to voip provider ?------? voip >> provider is freeswitch server ---? my carrier on freeswitch?. >> >> ? >> >> On the other way it?s work >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [8] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [9]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 21:28 >> >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> Question: what is the topology here? What is the phone and what is >> the carrier? And which leg needs to have the INFO DTMFs? >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:48 AM, Erwan Hermouet >> wrote: >> >> Ok i remove it >> >> ? >> >> But the samedi >> >> http://pastebin.freeswitch.org/20981 [11] >> >> ? >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [12] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [13]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 20:40 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> I think this might not be helpful: >> >> EXECUTE sofia/internal/12345 at bluetelconnect.fr [14]:5060 >> start_dtmf() >> >> Thats for doing inband DTMF. I would remove that altogether. >> >> -MC >> >> ? >> >> On Tue, May 28, 2013 at 11:29 AM, Erwan Hermouet >> wrote: >> >> Now i don?t see dtmf???????????:?( >> >> http://pastebin.freeswitch.org/20980 [16] >> >> ? >> >> show my pastebin >> >> ? >> >> tks mike >> >> ? >> >> DE?: freeswitch-users-bounces at lists.freeswitch.org [17] >> [mailto:freeswitch-users-bounces at lists.freeswitch.org [18]] DE LA >> PART DE Michael Collins >> ENVOY??: mardi 28 mai 2013 19:08 >> ??: FreeSWITCH Users Help >> OBJET?: Re: [Freeswitch-users] DTMF outbound call >> >> ? >> >> ? >> >> ? >> >> On Tue, May 28, 2013 at 12:07 AM, wrote: >> >> >> Tks Mike >> >> we are try with info >> >> on log i see that >> 2013-05-28 09:04:20.545789 [DEBUG] switch_rtp.c:1975 Send start >> packet >> for [1] ts=141083 dur=160/160/2240 seq=62753 >> 2013-05-28 09:04:20.527806 [DEBUG] switch_rtp.c:2902 RTP RECV DTMF >> 1:2240 >> >> but its not working again :( >> >> tks >> >> ? >> >> Call logs? >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [20] >> http://www.ClueCon.com [21] >> http://www.OSTAG.org [22] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [23] >> http://www.freeswitchsolutions.com [24] >> >> >> [25] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [26] >> http://wiki.freeswitch.org [27] >> http://www.cluecon.com [28] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [29] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [30] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [31] >> http://www.freeswitch.org [32] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [33] >> http://www.ClueCon.com [34] >> http://www.OSTAG.org [35] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [36] >> http://www.freeswitchsolutions.com [37] >> >> >> [38] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [39] >> http://wiki.freeswitch.org [40] >> http://www.cluecon.com [41] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [42] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [43] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [44] >> http://www.freeswitch.org [45] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [46] >> http://www.ClueCon.com [47] >> http://www.OSTAG.org [48] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [49] >> http://www.freeswitchsolutions.com [50] >> >> >> [51] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [52] >> http://wiki.freeswitch.org [53] >> http://www.cluecon.com [54] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [55] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [56] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [57] >> http://www.freeswitch.org [58] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [59] >> http://www.ClueCon.com [60] >> http://www.OSTAG.org [61] >> >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [62] >> http://www.freeswitchsolutions.com [63] >> >> >> [64] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [65] >> http://wiki.freeswitch.org [66] >> http://www.cluecon.com [67] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [68] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [69] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [70] >> http://www.freeswitch.org [71] >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org [72] >> http://www.ClueCon.com [73] >> http://www.OSTAG.org [74] >> > > _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org [75] >> http://www.freeswitchsolutions.com [76] >> >> >> [77] >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org [78] >> http://wiki.freeswitch.org [79] >> http://www.cluecon.com [80] >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org [81] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [82] >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> [83] >> http://www.freeswitch.org [84] > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org [86] > http://www.ClueCon.com [87] > http://www.OSTAG.org [88] > > > > Links: > ------ > [1] http://Bluetelconnect.fr/tcpdumpdtmf.log > [2] mailto:freeswitch-users-bounces at lists.freeswitch.org > [3] mailto:freeswitch-users-bounces at lists.freeswitch.org > [4] mailto:ehermouet at bluetel.fr > [5] mailto:freeswitch-users-bounces at lists.freeswitch.org > [6] mailto:freeswitch-users-bounces at lists.freeswitch.org > [7] mailto:ehermouet at bluetel.fr > [8] mailto:freeswitch-users-bounces at lists.freeswitch.org > [9] mailto:freeswitch-users-bounces at lists.freeswitch.org > [10] mailto:ehermouet at bluetel.fr > [11] http://pastebin.freeswitch.org/20981 > [12] mailto:freeswitch-users-bounces at lists.freeswitch.org > [13] mailto:freeswitch-users-bounces at lists.freeswitch.org > [14] http://bluetelconnect.fr > [15] mailto:ehermouet at bluetel.fr > [16] http://pastebin.freeswitch.org/20980 > [17] mailto:freeswitch-users-bounces at lists.freeswitch.org > [18] mailto:freeswitch-users-bounces at lists.freeswitch.org > [19] mailto:ehermouet at bluetel.fr > [20] http://www.FreeSWITCH.org > [21] http://www.ClueCon.com > [22] http://www.OSTAG.org > [23] mailto:consulting at freeswitch.org > [24] http://www.freeswitchsolutions.com > [25] > [26] http://www.freeswitch.org > [27] http://wiki.freeswitch.org > [28] http://www.cluecon.com > [29] mailto:FreeSWITCH-users at lists.freeswitch.org > [30] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [31] http://lists.freeswitch.org/mailman/options/freeswitch-users > [32] http://www.freeswitch.org > [33] http://www.FreeSWITCH.org > [34] http://www.ClueCon.com > [35] http://www.OSTAG.org > [36] mailto:consulting at freeswitch.org > [37] http://www.freeswitchsolutions.com > [38] > [39] http://www.freeswitch.org > [40] http://wiki.freeswitch.org > [41] http://www.cluecon.com > [42] mailto:FreeSWITCH-users at lists.freeswitch.org > [43] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [44] http://lists.freeswitch.org/mailman/options/freeswitch-users > [45] http://www.freeswitch.org > [46] http://www.FreeSWITCH.org > [47] http://www.ClueCon.com > [48] http://www.OSTAG.org > [49] mailto:consulting at freeswitch.org > [50] http://www.freeswitchsolutions.com > [51] > [52] http://www.freeswitch.org > [53] http://wiki.freeswitch.org > [54] http://www.cluecon.com > [55] mailto:FreeSWITCH-users at lists.freeswitch.org > [56] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [57] http://lists.freeswitch.org/mailman/options/freeswitch-users > [58] http://www.freeswitch.org > [59] http://www.FreeSWITCH.org > [60] http://www.ClueCon.com > [61] http://www.OSTAG.org > [62] mailto:consulting at freeswitch.org > [63] http://www.freeswitchsolutions.com > [64] > [65] http://www.freeswitch.org > [66] http://wiki.freeswitch.org > [67] http://www.cluecon.com > [68] mailto:FreeSWITCH-users at lists.freeswitch.org > [69] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [70] http://lists.freeswitch.org/mailman/options/freeswitch-users > [71] http://www.freeswitch.org > [72] http://www.FreeSWITCH.org > [73] http://www.ClueCon.com > [74] http://www.OSTAG.org > [75] mailto:consulting at freeswitch.org > [76] http://www.freeswitchsolutions.com > [77] > [78] http://www.freeswitch.org > [79] http://wiki.freeswitch.org > [80] http://www.cluecon.com > [81] mailto:FreeSWITCH-users at lists.freeswitch.org > [82] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > [83] http://lists.freeswitch.org/mailman/options/freeswitch-users > [84] http://www.freeswitch.org > [85] mailto:ehermouet at bluetel.fr > [86] http://www.FreeSWITCH.org > [87] http://www.ClueCon.com > [88] http://www.OSTAG.org From nandy1925 at gmail.com Wed May 29 13:00:08 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 29 May 2013 17:00:08 +0800 Subject: [Freeswitch-users] Best Installation MOD_GSMOPEN In-Reply-To: References: Message-ID: Hi Giovanni, Your welcome. I edited the Wiki page. Best rgds, Nandy ================================================ www.magicbox.ph - *the better magic* VoIP phone for Filipinos *Lapulapu City, Phils Phone: +63-32-3401807, Mobile: +63-920-6373450 *USA# :* (646)547-1226 *Worldwide:* [**462 + 17772930540 + #*] via any 200+ Access Numbers (37 Countries) On Wed, May 29, 2013 at 3:18 PM, Giovanni Maruzzelli wrote: > Thanks Nandy! > > > > > On Tue, May 28, 2013 at 4:34 AM, Nandy Dagondon wrote: > >> After wresting the problem installing gsmopen on Centos 6.4 x_64 (** >> libctb-2.6.so problem), I feel this information should be added in the >> Wiki in the pre-requisites >> >> ldconfig <-- after installing gsm library >> ---- Add this --- >> Check if the new library is included >> # ldconfig -p | grep gsm >> >> If no entries found, correct this problem by using the ldconfig examples >> at http://linux.101hacks.com/unix/ldconfig/ >> >> Run ldconfig again >> ----- End --- >> >> In my case, I added this rule file in /etc/ld.so.d directory >> # local-libs.conf >> /usr/local/lib >> >> >> /nandy >> >> >> >> On Tue, Feb 26, 2013 at 2:10 AM, Josue Diaz Cruz wrote: >> >>> ** >>> Good Afternoon. >>> >>> I beg your pardon if my question is really from a newbee. >>> >>> I am trying to install gsmopen in a system. I did it with CentOS 6 X64, >>> Debian 6, Ubuntu 12.10. Can some one provide me the best way to do it. >>> Cause i allways have problems in one point of the installation. Depending >>> the distro could it be one or other. I was thinking that could be nice to >>> give a complete requierements (libs) to install before do anything. >>> >>> If some one can provide me a easy, simple and not troubles, could be >>> nice: >>> >>> * System and version >>> * List of requierements. >>> * Better steps to avoid issues with libraries and compile. >>> >>> What i tried: >>> >>> Centos 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues when >>> try to load mod_gsmopen.so) >>> Debian 6 x86_64 --- Huawey E169 unlocked and voice active. (Issues with >>> the ttyUSB port) >>> Ubuntu 12.10 --- Huawey E169 unlocked and voice active. (Issues with >>> the ttyUSB port) >>> >>> E169 works perfectly. I have a mobile partner with voicecall and i test >>> it calling my land line without issues. >>> >>> >>> Thank you very much for your consideration. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/eaf72a5d/attachment.html From nandy1925 at gmail.com Wed May 29 13:13:55 2013 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 29 May 2013 17:13:55 +0800 Subject: [Freeswitch-users] IVR EXTERNAL TIMEOUT BRIDGE PROBLEM In-Reply-To: References: Message-ID: Hi, It looks the IVR timeout and it executes transfer to extension 1001 as expected in your dialplan. There's no exit sound file played though. What should be the expected of Exit? /nandy On Fri, Apr 12, 2013 at 2:18 AM, Diego Mendieta wrote: > > > ------------------------------ > From: d_iego at msn.com > To: freeswitch-users-owner at lists.freeswitch.org > Subject: IVR EXTERNAL TIMEOUT BRIDGE PROBLEM > Date: Thu, 11 Apr 2013 18:08:51 +0000 > > Hello World! > > I've been trying to solve an issue with my freeswitch pbx, please help me. > > I have set up a Welcome IVR for all incoming calls. Its seems to work > perfectley when dialed from internal extentions. The problems is that when > I dial from an external phone the Exit Action is not executed, instead the > phone call is terminated. > > I am using fusionpbx. Below I paste the log for both cases: > > EXECUTE sofia/external/76xxxx99 at 200.119.223.228 sleep(2000) > 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/76xxxx99 at 200.119.223.228 [BREAK] > 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/76xxxx99 at 200.119.223.228 [BREAK] > 2013-04-11 13:49:11.729600 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/76xxxx99 at 200.119.223.228 [BREAK] > 2013-04-11 13:49:11.749600 [DEBUG] sofia.c:5431 Channel sofia/external/ > 76xxxx99 at 200.119.223.228 entering state [ready][200] > EXECUTE sofia/external/76xxxx99 at 200.119.223.228 ivr(bienvenida) > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-exit' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-sub' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-exec-app' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-play-sound' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-back' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-top' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:796 building menu > 'bienvenida' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '1' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '2' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '3' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '/(^\d{3,6}$)/' > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_menu.c:433 Executing IVR > menu bienvenida > 2013-04-11 13:49:13.509600 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2013-04-11 13:49:13.549600 [DEBUG] switch_rtp.c:3204 Correct ip/port > confirmed. > 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_play_say.c:1672 done playing > file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav > 2013-04-11 13:49:29.749600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 > digits t/o 2000 > 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:395 digits '' > 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts > 2013-04-11 13:49:29.769600 [DEBUG] switch_ivr_menu.c:599 exit-sound > '(null)' > 2013-04-11 13:49:29.769600 [NOTICE] switch_core_state_machine.c:226 > sofia/external/76xxxx99 at 200.119.223.228 has executed the last dialplan > instruction, hanging up. > 2013-04-11 13:49:29.769600 [DEBUG] switch_channel.c:2846 (sofia/external/ > 76xxxx99 at 200.119.223.228) Callstate Change ACTIVE -> HANGUP > > internal call: > EXECUTE sofia/internal/101 at 192.168.0.26:5060 sleep(1000) > EXECUTE sofia/internal/101 at 192.168.0.26:5060set(hangup_after_bridge=true) > 2013-04-11 14:01:38.948600 [DEBUG] mod_dptools.c:1286 sofia/internal/ > 101 at 192.168.0.26:5060 SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/101 at 192.168.0.26:5060 ivr(bienvenida) > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-exit' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-sub' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-exec-app' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-play-sound' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-back' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:665 > switch_ivr_menu_stack_xml_add binding 'menu-top' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:796 building menu > 'bienvenida' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '1' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '2' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '3' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '8' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '9' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:857 binding menu > action 'menu-exec-app' to '/(^\d{3,6}$)/' > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_menu.c:433 Executing IVR > menu bienvenida > 2013-04-11 14:01:38.948600 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_play_say.c:1672 done playing > file C:\Program Files\FusionPBX\sounds/es/mx/maria/ivr/bienvenida.wav > 2013-04-11 14:01:55.188600 [DEBUG] switch_ivr_menu.c:348 waiting for 5/5 > digits t/o 2000 > 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:395 digits '' > 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:446 Maximum timeouts > 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_menu.c:599 exit-sound > '(null)' > EXECUTE sofia/internal/101 at 192.168.0.26:5060 bridge(sofia/internal/ > 1001 at 192.168.0.26) > 2013-04-11 14:01:55.208600 [DEBUG] switch_channel.c:1045 sofia/internal/ > 101 at 192.168.0.26:5060 EXPORTING[export_vars] [domain_name]=[192.168.0.26] > to event > 2013-04-11 14:01:55.208600 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2013-04-11 14:01:55.208600 [NOTICE] switch_channel.c:924 New Channel > sofia/internal/1001 at 192.168.0.26 [c5904e71-b9ab-4f60-a098-9c8280f38aa7] > > Below I paste my dial plan configuration: > > > > > > > > > > > > > Please help me. > > thank you, > > Diego > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/4ce2df51/attachment-0001.html From a.afzali2003 at gmail.com Wed May 29 13:14:33 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 29 May 2013 13:44:33 +0430 Subject: [Freeswitch-users] Encrypted RFC2833 DTMF Message-ID: Hi Guys, I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 dtmf digits. My voip provider uses a Huawei switch and does ask me to enable dtmf encryption but I could not find such parameter in sofia.xml.conf file. Appreciate all comments Afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/0a3ab425/attachment.html From gmangudai at gmail.com Wed May 29 15:02:29 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Wed, 29 May 2013 19:02:29 +0800 Subject: [Freeswitch-users] how to monitor a call session? Message-ID: can i monitor an ongoing call session by receiving freeswitch events through esl outbound socket? for example, i can have another process monitoring the sip call sessioin by acquiring call events(answer, hangup, etc.) from freeswitch. are the following steps ok? 1. create a socket server in a process 2. adding to the dialplan xml. 3. get the outbound socket connection from freeswitch 4. acquire call events should i do some event subscription before step 4? any response is appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/0b8cd4d8/attachment.html From gmangudai at gmail.com Wed May 29 15:03:04 2013 From: gmangudai at gmail.com (Vincent Xia) Date: Wed, 29 May 2013 19:03:04 +0800 Subject: [Freeswitch-users] is it possible to disable hold function Message-ID: is it possible to disable hold function of a given extension by modifying freeswitch config, e.g. dialplan\default.xml? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/b3c9d3c9/attachment.html From mike at jerris.com Wed May 29 16:09:41 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 29 May 2013 08:09:41 -0400 Subject: [Freeswitch-users] Freeswitch forks new processes? In-Reply-To: References: Message-ID: <9F7170F8-CAA7-4279-9472-C152555DA780@jerris.com> Freeswitch doesn't fork like that ever. On May 29, 2013, at 12:14 AM, Yehavi Bourvine wrote: > Hello, > > I have a problem with 1.2.9 (still presented in GIT head from last week); under 40-50 sessions load, incoming sessions starts getting hung in RINGING state. I did not open a JIRA yet as it does not happens when DEBUG level is active, but I noticed a strange behaviour: when it happens I see more than one freeswitch process. During normal work I have only one. > > When does freeswitch creates more than one process? BTW, I run it with -nc -nonat From a.mykhalkiv at kwebbl.com Wed May 29 09:23:12 2013 From: a.mykhalkiv at kwebbl.com (Anatolii) Date: Tue, 28 May 2013 22:23:12 -0700 (PDT) Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: References: <1369661283368-7591135.post@n2.nabble.com> Message-ID: <1369804992163-7591213.post@n2.nabble.com> Thanks for your answer. Yesterday, I try another way. I add tiers and agents in dialplan. For example, < action application="set" data="res=${callcenter_config(agent add agent_name agent_type" /> or is it a bad choice? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-callcenter-configuration-don-t-register-agents-and-tiers-tp7591135p7591213.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed May 29 16:14:55 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 29 May 2013 08:14:55 -0400 Subject: [Freeswitch-users] Encrypted RFC2833 DTMF In-Reply-To: References: Message-ID: <2620FFA7-1362-4255-9FA2-1BAD96AFBDEE@jerris.com> We support SRTP, which encrypts the entire media stream including the 2833 digits. I'm not familiar with a standard that just encrypts the dtmf. Did they provide any details of what this standard is? Mike On May 29, 2013, at 5:14 AM, afshin afzali wrote: > Hi Guys, > > I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 dtmf digits. My voip provider uses a Huawei switch and does ask me to enable dtmf encryption but I could not find such parameter in sofia.xml.conf file. > From mike at jerris.com Wed May 29 16:15:43 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 29 May 2013 08:15:43 -0400 Subject: [Freeswitch-users] is it possible to disable hold function In-Reply-To: References: Message-ID: <73B269CF-BDAE-45EE-9E34-58F2D431CED5@jerris.com> http://wiki.freeswitch.org/wiki/Variable_disable_hold On May 29, 2013, at 7:03 AM, Vincent Xia wrote: > is it possible to disable hold function of a given extension by modifying freeswitch config, e.g. dialplan\default.xml? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/d8a5c32e/attachment.html From bdfoster at davri.com Wed May 29 16:37:14 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 08:37:14 -0400 Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: <1369804992163-7591213.post@n2.nabble.com> References: <1369661283368-7591135.post@n2.nabble.com> <1369804992163-7591213.post@n2.nabble.com> Message-ID: It depends on what you are tryibg to accomplish and the complexity of your mod_callcenter setup. If you want your agents to be able to log in and log out themselves, doing it in the dial plan by giving them an access number is great. You could use params in the users entry in directory to set tier and queue as well, and set those as part of your login dialplan. Just an idea. My approach is to use xml dialplan as much as possible. There are a few reasons why I do that and certainly doesn't apply to everyone's use case. I saw this yesterday and I was going to add maybe putting together a simple web ui for your agents to use for this purpose. No special databases, nothing too spectacular. Use the users id and vm password for credentials. I'm assuming yoh arent using fusionpbx that has similar functiinality already built in. - BDF On May 29, 2013 8:17 AM, "Anatolii" wrote: > Thanks for your answer. > > Yesterday, I try another way. > I add tiers and agents in dialplan. For example, > < action application="set" data="res=${callcenter_config(agent add > agent_name agent_type" /> > > or is it a bad choice? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Problem-with-callcenter-configuration-don-t-register-agents-and-tiers-tp7591135p7591213.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://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/20130529/cdb325db/attachment.html From bdfoster at davri.com Wed May 29 16:40:25 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 08:40:25 -0400 Subject: [Freeswitch-users] Problem with callcenter configuration (don't register agents and tiers) In-Reply-To: References: <1369661283368-7591135.post@n2.nabble.com> <1369804992163-7591213.post@n2.nabble.com> Message-ID: Sorry, apparently I did not read your dialplan before posting. It's probably a good idea NOT to add agents via dialplan. Use static agents, set them inside the callcenter.conf.xml assuming agents dont change much. Use agent status to login/log out of the queues. You can change status of agents in a similar fashion inside the dialplan. - BDF On May 29, 2013 8:37 AM, "Brian Foster" wrote: > It depends on what you are tryibg to accomplish and the complexity of your > mod_callcenter setup. If you want your agents to be able to log in and log > out themselves, doing it in the dial plan by giving them an access number > is great. You could use params in the users entry in directory to set tier > and queue as well, and set those as part of your login dialplan. Just an > idea. > > My approach is to use xml dialplan as much as possible. There are a few > reasons why I do that and certainly doesn't apply to everyone's use case. > > I saw this yesterday and I was going to add maybe putting together a > simple web ui for your agents to use for this purpose. No special > databases, nothing too spectacular. Use the users id and vm password for > credentials. I'm assuming yoh arent using fusionpbx that has similar > functiinality already built in. > > - BDF > On May 29, 2013 8:17 AM, "Anatolii" wrote: > >> Thanks for your answer. >> >> Yesterday, I try another way. >> I add tiers and agents in dialplan. For example, >> < action application="set" data="res=${callcenter_config(agent add >> agent_name agent_type" /> >> >> or is it a bad choice? >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-callcenter-configuration-don-t-register-agents-and-tiers-tp7591135p7591213.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://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/20130529/8f116d05/attachment-0001.html From peter at olssononline.se Wed May 29 16:44:49 2013 From: peter at olssononline.se (Peter Olsson) Date: Wed, 29 May 2013 14:44:49 +0200 Subject: [Freeswitch-users] how to monitor a call session? In-Reply-To: References: Message-ID: What you describe here will also force you to control the flow of the call, and if I understand you correctly you only want to monitor events etc. In that case it's much better to use inbound ESL connection. However, you need to login, and "subscribe" to the events you're interested in. Read more here: http://wiki.freeswitch.org/wiki/Mod_event_socket#Inbound /Peter 2013/5/29 Vincent Xia > can i monitor an ongoing call session by receiving freeswitch events > through esl outbound socket? for example, i can have another process > > monitoring the sip call sessioin by acquiring call events(answer, hangup, > etc.) from freeswitch. > > are the following steps ok? > 1. create a socket server in a process > 2. adding to the > dialplan xml. > 3. get the outbound socket connection from freeswitch > 4. acquire call events > > should i do some event subscription before step 4? > > any response is appreciated! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/1dd3066a/attachment.html From bdfoster at davri.com Wed May 29 16:46:12 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 08:46:12 -0400 Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error In-Reply-To: <1369813546562-7591226.post@n2.nabble.com> References: <1369813546562-7591226.post@n2.nabble.com> Message-ID: I haven't interfaced with a Sonus SBC before (knowingly) but haven't received the file requested from the OP yet. Go ahead and post a console log with global siptrace on into pastebin.freeswitch.org and post back with a link. Let's see if we can get this figured out. - BDF On May 29, 2013 3:50 AM, "arag00rn" wrote: > Hello Brian, > > did you manage to solve this issue ? > I have exactly the same problem. > > FreeSWITCH (Version 1.5.1b git 16690e4 2013-05-09 19:05:09Z) > > The external SBC is a Sonus. > > BR, > Andrea > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591226.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://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/20130529/f9e2649d/attachment.html From nneul at mst.edu Wed May 29 17:07:12 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Wed, 29 May 2013 08:07:12 -0500 Subject: [Freeswitch-users] Freeswitch forks new processes? In-Reply-To: <9F7170F8-CAA7-4279-9472-C152555DA780@jerris.com> References: <9F7170F8-CAA7-4279-9472-C152555DA780@jerris.com> Message-ID: <51A5FD80.604@mst.edu> It forks if you use any 'system' commands, or send mail from voicemail, or similar. Are you using any of those types of functions? -- Nathan On 05/29/2013 07:09 AM, Michael Jerris wrote: > Freeswitch doesn't fork like that ever. > > On May 29, 2013, at 12:14 AM, Yehavi Bourvine wrote: > >> Hello, >> >> I have a problem with 1.2.9 (still presented in GIT head from last week); under 40-50 sessions load, incoming sessions starts getting hung in RINGING state. I did not open a JIRA yet as it does not happens when DEBUG level is active, but I noticed a strange behaviour: when it happens I see more than one freeswitch process. During normal work I have only one. >> >> When does freeswitch creates more than one process? BTW, I run it with -nc -nonat > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From anthony.minessale at gmail.com Wed May 29 17:30:42 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 May 2013 08:30:42 -0500 Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error In-Reply-To: References: <1369813546562-7591226.post@n2.nabble.com> Message-ID: After you update to head, of course. On May 29, 2013 7:50 AM, "Brian Foster" wrote: > I haven't interfaced with a Sonus SBC before (knowingly) but haven't > received the file requested from the OP yet. Go ahead and post a console > log with global siptrace on into pastebin.freeswitch.org and post back > with a link. Let's see if we can get this figured out. > > - BDF > On May 29, 2013 3:50 AM, "arag00rn" wrote: > >> Hello Brian, >> >> did you manage to solve this issue ? >> I have exactly the same problem. >> >> FreeSWITCH (Version 1.5.1b git 16690e4 2013-05-09 19:05:09Z) >> >> The external SBC is a Sonus. >> >> BR, >> Andrea >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591226.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://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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/ca2c1ddf/attachment.html From a.afzali2003 at gmail.com Wed May 29 17:39:36 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 29 May 2013 18:09:36 +0430 Subject: [Freeswitch-users] Encrypted RFC2833 DTMF In-Reply-To: <2620FFA7-1362-4255-9FA2-1BAD96AFBDEE@jerris.com> References: <2620FFA7-1362-4255-9FA2-1BAD96AFBDEE@jerris.com> Message-ID: Hi Mike, Actually I'm not. This is a request from my local operator. As you said I told them that is possible to encrypt entire rtp stream by SRTP / ZRTP techniques. She says me that (on Huawei Switch) the encryption option just there when she selects rfc2833 for digit handling and not on entire media stream !!! Thank you so much, Afshin On Wed, May 29, 2013 at 4:44 PM, Michael Jerris wrote: > We support SRTP, which encrypts the entire media stream including the 2833 > digits. I'm not familiar with a standard that just encrypts the dtmf. Did > they provide any details of what this standard is? > > Mike > > On May 29, 2013, at 5:14 AM, afshin afzali wrote: > > > Hi Guys, > > > > I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 > dtmf digits. My voip provider uses a Huawei switch and does ask me to > enable dtmf encryption but I could not find such parameter in > sofia.xml.conf file. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/98fd2a71/attachment-0001.html From eoaddai at gmail.com Wed May 29 16:41:45 2013 From: eoaddai at gmail.com (eoaddai) Date: Wed, 29 May 2013 05:41:45 -0700 (PDT) Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION In-Reply-To: References: <1369468901841-7591099.post@n2.nabble.com> Message-ID: HI, i actually changed the goip8 to goip4 gateway, and the samme error SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION is been generated. Anyone with any ideas too? On 27 May 2013 19:01, Steven Ayre [via freeswitch-users] < ml-node+s2379917n7591155h78 at n2.nabble.com> wrote: > recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >> ------------------------------------------------------------ >> ------------ >> SIP/2.0 606 Not Acceptable > > > Yes. The 606 packet is generated by the goip and sent by the goip to > freeswitch. > > We/FreeSWITCH cannot tell you why the goip sent 606, you would need to > check its logs/records or with the provider/manufacturer to see why. > > The only thing I would say is a codec mismatch is normally 488 not 606, so > chances are it's something else... but that is assuming that they are > following the standards, which is not necessarily the case. > > One thing I would try is setting the verbose_sdp variable: > > See http://wiki.freeswitch.org/wiki/Variable_verbose_sdp for > > > -Steve > > > > > On 26 May 2013 12:20, eoaddai <[hidden email] > > wrote: > >> So, is the call hitting the goip at all? or, freeswitch is unable to >> process it to the goip? >> >> >> On 26 May 2013 10:32, Daniel Ivanov [via freeswitch-users] <[hidden >> email] > wrote: >> >>> Maybe you're not sending them the right codecs or trying to run a >>> feature they don't have. Revisit your vars.xml and sip_profiles .xml. . >>> Ultimately contact the provider to ask them what youre doing wrong in your >>> sdps. >>> On May 26, 2013 11:26 AM, "eoaddai" <[hidden email]> >>> wrote: >>> >>>> Hi, my first time of posting stuff here. I really need help. >>>> I keep on not getting my calls go through freeswitch to my goip >>>> gateway. The >>>> following is the freeswitch log with siptrace turned on i get. Please >>>> help >>>> me: >>>> >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing >>>> global variables >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [origination_caller_id_number]=[0264370536] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_from]=[0264370536] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_to]=[0267577771] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_app]=[true] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [originate_timeout]=[60] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>> [ignore_early_media]=[true] >>>> 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel >>>> sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] >>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 >>>> (sofia/external/0267577771) State Change CS_NEW -> CS_INIT >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0267577771) Running State Change CS_INIT >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>>> (sofia/external/0267577771) State INIT >>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 >>>> sofia/external/0267577771 >>>> SOFIA INIT >>>> 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: >>>> v=0 >>>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.50.1 >>>> t=0 0 >>>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 >>>> (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>>> (sofia/external/0267577771) State INIT going to sleep >>>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE [hidden email]SIP/2.0 >>>> >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>> Max-Forwards: 70 >>>> From: "" <[hidden email] >>>> >;tag=DXjN0tK6mcc5S >>>> To: <[hidden email] >>>> > >>>> >>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 INVITE >>>> Contact: >>>> User-Agent: >>>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>>> REGISTER, REFER, NOTIFY >>>> Supported: timer, precondition, path, replaces >>>> Allow-Events: talk, hold, conference, refer >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 201 >>>> X-FS-Support: update_display,send_info >>>> Remote-Party-ID: >>>> <[hidden email] >>>> >;party=calling;screen=yes;privacy=off >>>> >>>> >>>> v=0 >>>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.50.1 >>>> t=0 0 >>>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> >>>> ------------------------------------------------------------------------ >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0267577771) Running State Change CS_ROUTING >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>>> (sofia/external/0267577771) State ROUTING >>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 >>>> sofia/external/0267577771 >>>> SOFIA ROUTING >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 >>>> (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>>> (sofia/external/0267577771) State ROUTING going to sleep >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>>> (sofia/external/0267577771) State CONSUME_MEDIA >>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>>> (sofia/external/0267577771) State CONSUME_MEDIA going to sleep >>>> 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel >>>> sofia/external/0267577771 entering state [calling][0] >>>> recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 606 Not Acceptable >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>> From: "" <[hidden email] >>>> >;tag=DXjN0tK6mcc5S >>>> To: <[hidden email] >>>> >;tag=1662509363 >>>> >>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 INVITE >>>> User-Agent: dble >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: >>>> >>>> ------------------------------------------------------------------------ >>>> ACK [hidden email]SIP/2.0 >>>> >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>> Max-Forwards: 70 >>>> From: "" <[hidden email] >>>> >;tag=DXjN0tK6mcc5S >>>> To: <[hidden email] >>>> >;tag=1662509363 >>>> >>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 ACK >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel >>>> sofia/external/0267577771 entering state [terminated][606] >>>> 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup >>>> sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 Originate >>>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal >>>> sofia/external/0267577771 [KILL] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0267577771) Running State Change CS_HANGUP >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>>> (sofia/external/0267577771) State HANGUP >>>> 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel >>>> sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 >>>> sofia/external/0267577771 Standard HANGUP, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>>> (sofia/external/0267577771) State HANGUP going to sleep >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 >>>> (sofia/external/0267577771) Callstate Change DOWN -> HANGUP >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 >>>> (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0267577771) Running State Change CS_REPORTING >>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 >>>> (sofia/external/0267577771) State REPORTING >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 >>>> sofia/external/0267577771 Standard REPORTING, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 >>>> (sofia/external/0267577771) State REPORTING going to sleep >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0267577771 [BREAK] >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 >>>> (sofia/external/0267577771) Locked, Waiting on external entities >>>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session 3 >>>> (sofia/external/0267577771) Ended >>>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close >>>> Channel >>>> sofia/external/0267577771 [CS_DESTROY] >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 >>>> (sofia/external/0267577771) Callstate Change HANGUP -> DOWN >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 >>>> (sofia/external/0267577771) Running State Change CS_DESTROY >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>>> (sofia/external/0267577771) State DESTROY >>>> 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 >>>> sofia/external/0267577771 >>>> SOFIA DESTROY >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 >>>> sofia/external/0267577771 Standard DESTROY >>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>>> (sofia/external/0267577771) State DESTROY going to sleep >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing >>>> global variables >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [origination_caller_id_number]=[0264370536] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_from]=[0264370536] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_to]=[0249230704] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [plivo_app]=[true] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [originate_timeout]=[60] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>> [ignore_early_media]=[true] >>>> 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel >>>> sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] >>>> 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 >>>> (sofia/external/0249230704) State Change CS_NEW -> CS_INIT >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0249230704) Running State Change CS_INIT >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>>> (sofia/external/0249230704) State INIT >>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 >>>> sofia/external/0249230704 >>>> SOFIA INIT >>>> 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: >>>> v=0 >>>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.50.1 >>>> t=0 0 >>>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> a=sendrecv >>>> >>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 >>>> (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>>> (sofia/external/0249230704) State INIT going to sleep >>>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: >>>> >>>> ------------------------------------------------------------------------ >>>> INVITE [hidden email]SIP/2.0 >>>> >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>> Max-Forwards: 70 >>>> From: "" <[hidden email] >>>> >;tag=e6Be2N49HN2QN >>>> To: <[hidden email] >>>> > >>>> >>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 INVITE >>>> Contact: >>>> User-Agent: >>>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>>> REGISTER, REFER, NOTIFY >>>> Supported: timer, precondition, path, replaces >>>> Allow-Events: talk, hold, conference, refer >>>> Content-Type: application/sdp >>>> Content-Disposition: session >>>> Content-Length: 201 >>>> X-FS-Support: update_display,send_info >>>> Remote-Party-ID: >>>> <[hidden email] >>>> >;party=calling;screen=yes;privacy=off >>>> >>>> >>>> v=0 >>>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>>> s=FreeSWITCH >>>> c=IN IP4 10.10.50.1 >>>> t=0 0 >>>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>>> a=rtpmap:101 telephone-event/8000 >>>> a=fmtp:101 0-16 >>>> a=ptime:20 >>>> >>>> ------------------------------------------------------------------------ >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0249230704) Running State Change CS_ROUTING >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>>> (sofia/external/0249230704) State ROUTING >>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 >>>> sofia/external/0249230704 >>>> SOFIA ROUTING >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 >>>> (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>>> (sofia/external/0249230704) State ROUTING going to sleep >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>>> (sofia/external/0249230704) State CONSUME_MEDIA >>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>>> (sofia/external/0249230704) State CONSUME_MEDIA going to sleep >>>> 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel >>>> sofia/external/0249230704 entering state [calling][0] >>>> recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >>>> >>>> ------------------------------------------------------------------------ >>>> SIP/2.0 606 Not Acceptable >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>> From: "" <[hidden email] >>>> >;tag=e6Be2N49HN2QN >>>> To: <[hidden email] >>>> >;tag=633680086 >>>> >>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 INVITE >>>> User-Agent: dble >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: >>>> >>>> ------------------------------------------------------------------------ >>>> ACK [hidden email]SIP/2.0 >>>> >>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>> Max-Forwards: 70 >>>> From: "" <[hidden email] >>>> >;tag=e6Be2N49HN2QN >>>> To: <[hidden email] >>>> >;tag=633680086 >>>> >>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>> CSeq: 44398096 ACK >>>> Content-Length: 0 >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel >>>> sofia/external/0249230704 entering state [terminated][606] >>>> 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup >>>> sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal >>>> sofia/external/0249230704 [KILL] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0249230704) Running State Change CS_HANGUP >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>>> (sofia/external/0249230704) State HANGUP >>>> 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel >>>> sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 >>>> sofia/external/0249230704 Standard HANGUP, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>>> (sofia/external/0249230704) State HANGUP going to sleep >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 >>>> (sofia/external/0249230704) Callstate Change DOWN -> HANGUP >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 >>>> (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>>> (sofia/external/0249230704) Running State Change CS_REPORTING >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>>> (sofia/external/0249230704) State REPORTING >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 >>>> sofia/external/0249230704 Standard REPORTING, cause: >>>> INCOMPATIBLE_DESTINATION >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>>> (sofia/external/0249230704) State REPORTING going to sleep >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 >>>> (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>> signal >>>> sofia/external/0249230704 [BREAK] >>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 >>>> (sofia/external/0249230704) Locked, Waiting on external entities >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 Originate >>>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session 4 >>>> (sofia/external/0249230704) Ended >>>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close >>>> Channel >>>> sofia/external/0249230704 [CS_DESTROY] >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 >>>> (sofia/external/0249230704) Callstate Change HANGUP -> DOWN >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 >>>> (sofia/external/0249230704) Running State Change CS_DESTROY >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>>> (sofia/external/0249230704) State DESTROY >>>> 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 >>>> sofia/external/0249230704 >>>> SOFIA DESTROY >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 >>>> sofia/external/0249230704 Standard DESTROY >>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>>> (sofia/external/0249230704) State DESTROY going to sleep >>>> >>>> >>>> >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> [hidden email] >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> [hidden email] >>>> >>>> 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: >>> [hidden email] >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> [hidden email] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------ >>> If you reply to this email, your message will be added to the >>> discussion below: >>> >>> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591120.html >>> To unsubscribe from SIP/2.0 606 Not Acceptable and >>> INCOMPATIBLE_DESTINATION, click here. >>> NAML >>> >> >> >> >> -- >> *Emmanuel O. Addai,* >> *> target="_blank">+233(0)26 757 7771* >> >> ------------------------------ >> View this message in context: Re: SIP/2.0 606 Not Acceptable and >> INCOMPATIBLE_DESTINATION >> >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> 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: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591155.html > To unsubscribe from SIP/2.0 606 Not Acceptable and > INCOMPATIBLE_DESTINATION, click here > . > NAML > -- *Emmanuel O. Addai,* *+233(0)26 757 7771* -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591236.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/267ae88a/attachment-0001.html From alan.scales74 at gmail.com Wed May 29 17:03:02 2013 From: alan.scales74 at gmail.com (Alan Scales) Date: Wed, 29 May 2013 16:03:02 +0300 Subject: [Freeswitch-users] Transcoding issue (D150 card, 1.3.4.1 library) Message-ID: Hello, While setting up the environment needed to run the FreeSWITCH mod_sangoma_codec module, I've ran into a sngtc_server issue. For transcoding I'm using a D150 Sangoma transcoding card and the problem appears while starting the sngtc_server. Before starting the server, the card was configured with *sngtc_cfg*, and responded to pings. The module *detection* works, the card receives a *reset* command, and after the 20 seconds wait period, module *activation* fails with: Error: cOCTVC1_MAIN_MSG_MODULE_CLEANUP_API_RESOURCE_CID failed, rc = 0x0a000301 (cOCTVC1_GENERIC_RC_API_INVALID_CMD_LENGTH) After the failure, the procedure is repeated: { detect, reset, sleep, activate, fail } I suspect that the issue happens because of the fact that the *sngtc_server*and *card* firmwares do not match. However, after stopping the server, the firmware update (seems to be a downgrade) command also fails: Module [00-0c-90-1b-4d-6f] Upgrading Firmware cur=[01.06.02-B4-PR] to new=[01.04.03-B53-PR] Error: cOCTVC1_MAIN_MSG_FILE_OPEN_CID failed, rc = 0x0a010210 (cOCTVC1_MAIN_RC_FILE_MAX_WRITE_BYTE_SIZE) Regarding the logging, the only logs I've followed are in * /var/log/sngtc_server.log* Any hints on this matter would be highly appreciated, especially since it's not really a FreeSWITCH issue. Thank you, Alan Scales -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/22c164d3/attachment.html From bdfoster at davri.com Wed May 29 17:57:09 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 09:57:09 -0400 Subject: [Freeswitch-users] Encrypted RFC2833 DTMF In-Reply-To: References: <2620FFA7-1362-4255-9FA2-1BAD96AFBDEE@jerris.com> Message-ID: Your operator is likely talking about the DTMF 'method' (inband, SIP INFO, rfc2833), not encryption. In which case yoi are probably set up to do that already. Post a console log of a sample call on pastebin.freeswitch.org if yiu want is to confirm, and reply to this thread with a link. - BDF On May 29, 2013 9:45 AM, "afshin afzali" wrote: > Hi Mike, > > Actually I'm not. This is a request from my local operator. As you said I > told them that is possible to encrypt entire rtp stream by SRTP / ZRTP > techniques. She says me that (on Huawei Switch) the encryption option just > there when she selects rfc2833 for digit handling and not on entire media > stream !!! > Thank you so much, > > Afshin > > > On Wed, May 29, 2013 at 4:44 PM, Michael Jerris wrote: > >> We support SRTP, which encrypts the entire media stream including the >> 2833 digits. I'm not familiar with a standard that just encrypts the dtmf. >> Did they provide any details of what this standard is? >> >> Mike >> >> On May 29, 2013, at 5:14 AM, afshin afzali >> wrote: >> >> > Hi Guys, >> > >> > I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 >> dtmf digits. My voip provider uses a Huawei switch and does ask me to >> enable dtmf encryption but I could not find such parameter in >> sofia.xml.conf file. >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/bc685db4/attachment.html From B.Tietz at pinguin.ag Wed May 29 18:41:52 2013 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Wed, 29 May 2013 16:41:52 +0200 Subject: [Freeswitch-users] Errer Message-ID: <07BF4904977CC645B485E970424193AD130923A407@localhost> Hi, I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... FS 1.2.5.3 works! Dialplan is Anything I can try?! VG, Benjamin T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/5fed8438/attachment.html From admin at smallunix.net Wed May 29 18:43:40 2013 From: admin at smallunix.net (arag00rn) Date: Wed, 29 May 2013 07:43:40 -0700 (PDT) Subject: [Freeswitch-users] No audio RTP ports available! & I/O Error In-Reply-To: References: <1369813546562-7591226.post@n2.nabble.com> Message-ID: Hello Brian, sorry the reply was for jaganthoutam. Unfortunately, now I cannot make more test because it's a production ambient. I switched back to a stable version (Version 1.2.10 git e1a7734). I hope that next week I'll be more free to make more tests. BR, Andrea 2013/5/29 Brian Foster-2 [via freeswitch-users] < ml-node+s2379917n7591240h59 at n2.nabble.com> > I haven't interfaced with a Sonus SBC before (knowingly) but haven't > received the file requested from the OP yet. Go ahead and post a console > log with global siptrace on into pastebin.freeswitch.org and post back > with a link. Let's see if we can get this figured out. > > - BDF > On May 29, 2013 3:50 AM, "arag00rn" <[hidden email]> > wrote: > >> Hello Brian, >> >> did you manage to solve this issue ? >> I have exactly the same problem. >> >> FreeSWITCH (Version 1.5.1b git 16690e4 2013-05-09 19:05:09Z) >> >> The external SBC is a Sonus. >> >> BR, >> Andrea >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591226.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> 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: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591240.html > To unsubscribe from No audio RTP ports available! & I/O Error, click here > . > NAML > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-audio-RTP-ports-available-I-O-Error-tp7590933p7591246.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/aa6d698b/attachment-0001.html From msc at freeswitch.org Wed May 29 19:19:42 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 May 2013 08:19:42 -0700 Subject: [Freeswitch-users] Dialplan not executing on continue_on_fail=true In-Reply-To: References: <519F5CCD.8000609@mst.edu> <519F6087.3030706@mst.edu> Message-ID: I think Nathan is correct on this one. You don't have an explicit bridge app in your dialplan so there's nothing from which to continue. If you cannot use the bridge app and instead use the perl script then I'm pretty sure you're going to need to handle everything inside the script itself. -MC On Tue, May 28, 2013 at 9:58 PM, Ashish gautam wrote: > The originate string is like this: > > api originate > {voiceMessageID=$voiceMsgID,respreqd=$response_required,mobnum=$mobilenum,lang=$language,ignore_early_media=true,continue_on_fail=true}freetdm/1/a/$mobilenum > 47673501 XML public > > > > On Tue, May 28, 2013 at 10:35 PM, Michael Collins wrote: > >> What does your originate string look like? >> -MC >> >> >> On Sun, May 26, 2013 at 10:49 PM, Ashish gautam wrote: >> >>> Hi Nathan, >>> >>> Even setting the api_hangup_hook=perl hook.pl in the originate string >>> does not work. hook.pl does not get executed on hangup. It has to be >>> done some other way I guess. >>> >>> Thanks. >>> >>> >>> On Fri, May 24, 2013 at 6:13 PM, Nathan Neulinger wrote: >>> >>>> I don't think that's going to do what you want... (May be wrong.) >>>> >>>> I think that continue_on_fail is only going to apply to the rules for >>>> the received call on this extension, not the received call on the outgoing >>>> leg. >>>> >>>> i.e. there are no dialplan rules in effect for the outgoing call that >>>> you initiated, and that's where the failure is occurring. For these >>>> dialplan rules, I think the only failure would be if your IVR (I assume >>>> that's was ash.pl is) didn't answer. >>>> >>>> Like I said, not certain of this, maybe some else can chime in, but I >>>> think you're going to have to handle that failure as a part of your >>>> originate on the outbound call. Something like putting >>>> >>>> originate {api_hangup_hook=perl hook.pl}sofia/..... >>>> >>>> Where you cause the call to take place. >>>> >>>> -- Nathan >>>> >>>> >>>> On 05/24/2013 07:37 AM, Ashish gautam wrote: >>>> >>>>> I am generating an outgoing call through mod_event_socket and then >>>>> transferring it to this dialplan. >>>>> >>>>> On Fri, May 24, 2013 at 5:57 PM, Nathan Neulinger >>>> nneul at mst.edu>> wrote: >>>>> >>>>> I may be misunderstanding - but where are you causing it to ring a >>>>> device? >>>>> >>>>> You've told it to internally answer the call, and then not do >>>>> anything. There's no bridging to an actual extension. >>>>> >>>>> Only thing I see that would happen is it running perl/ash.pl < >>>>> http://ash.pl>, unclear if it would in term execute >>>>> hook.pl when that script finished (I don't know >>>>> what that behavior is expected to be). >>>>> >>>>> >>>>> -- Nathan >>>>> >>>>> >>>>> On 05/24/2013 07:17 AM, Ashish gautam wrote: >>>>> >>>>> Hi, >>>>> >>>>> I have a dialplan as follows: >>>>> >>>>> >>>>> >>>>> >>>> expression="^(47673501)$"> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> when the called party does not pick up the phone or is busy, >>>>> the dialplan does not proceed and hook.pl >>>>> >>>>> >>>>> >>>>> does not get executed. >>>>> >>>>> Please help >>>>> -- >>>>> Ashish Gautam >>>>> >>>>> IVR Developer >>>>> >>>>> Nucleus Microsystems (Pvt.) Ltd. >>>>> >>>>> >>>>> >>>>> -- >>>>> ------------------------------**__----------------------------**-- >>>>> Nathan Neulinger nneul at mst.edu >>>>> >>>>> Missouri S&T Information Technology (573) 612-1412 >>>>> System Administrator - Architect >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Ashish Gautam >>>>> >>>>> IVR Developer >>>>> >>>>> Nucleus Microsystems (Pvt.) Ltd. >>>>> >>>>> Ph. 011 47574758 >>>>> >>>> >>>> -- >>>> ------------------------------**------------------------------ >>>> Nathan Neulinger nneul at mst.edu >>>> Missouri S&T Information Technology (573) 612-1412 >>>> System Administrator - Architect >>>> >>> >>> >>> >>> -- >>> Ashish Gautam >>> >>> IVR Developer >>> >>> Nucleus Microsystems (Pvt.) Ltd. >>> >>> Ph. 011 47574758 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Ashish Gautam > > IVR Developer > > Nucleus Microsystems (Pvt.) Ltd. > > Ph. 011 47574758 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/cc38c3a0/attachment.html From msc at freeswitch.org Wed May 29 19:22:20 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 May 2013 08:22:20 -0700 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: It sounds like you have an ESL program that is not consuming events quickly enough and the event queue is filling up. -MC On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: > so setting the limit at 150 was fine but as soon as I set it to 200 I've > now hit a problem. > Freeswitch has slowly grown to 372 sessions and getting lots of these > errors in the cli > > 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more > events..... > > UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, 606 > microseconds > FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready > 1500 session(s) since startup > 372 session(s) - 0 out of max 10 per sec > 10000 session(s) max > min idle cpu 0.00/100.00 > Current Stack Size/Max 240K/8192K > > > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 15:01, jay binks wrote: > >> http://wiki.freeswitch.org/wiki/Limit >> >> >> >> >> On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >> >>> >>> Sorry for the thread hijack, but on the other hand, is it possible to >>> limit the number of outgoing CPS? Don't seem to see that either in the wiki >>> or a way to make it work. >>> >>> -- >>> Thanks, >>> Muhammad Naseer Bhatti >>> >>> >>> >>> Alex Ynema wrote: >>> >>> Cheers Avi I've now changed that to hash as I don't need it to be >>> persistent. >>> What should I see in the clie to confirm this is working without >>> attempting 150+ calls >>> >>> Basically I've added this to my default.xml >>> >>> >>> >> data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>> >>> >>> >>> >>> >> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>> >>> >>> >>> >>> >>> >> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>> >>> >>> >>> >>> *Alex Ynema** *| IT Consultant >>> alex at opensystems.net.au >>> >>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>> Office: +61 8 9427 2500 >>> Mobile: +61 404 796 894 >>> >>> IT Consultant for Open Systems Support >>> www.opensystems.net.au >>> >>> >>> On 29 May 2013 14:13, Avi Marcus wrote: >>> >>>> ... just note that's stored in a database (db) not ram (hash) so if you >>>> don't need to share it / have persistence, just store it in ram. >>>> >>>> -Avi >>>> >>>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >>>> >>>>> I've implemented this in default.xml hoping to limit each of my two >>>>> gateways to 150. >>>>> Based on what's in >>>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>>> hopefully that works. >>>>> >>>>> >>>>> >>>> expression="zetta-cisco-1"> >>>>> >>>>> >>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>> >>>>> >>>>> >>>>> >>>> expression="zetta-cisco-2"> >>>>> >>>>> >>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>> >>>>> >>>>> >>>>> >>>>> *Alex Ynema** *| IT Consultant >>>>> alex at opensystems.net.au >>>>> >>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>> Office: +61 8 9427 2500 >>>>> Mobile: +61 404 796 894 >>>>> >>>>> IT Consultant for Open Systems Support >>>>> www.opensystems.net.au >>>>> >>>>> >>>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>>> >>>>>> Hi, >>>>>> I'm trying to figure out how to limit the number of calls a Gateway >>>>>> is allowed to use. Our Sip provider has provided up with 200 which I need >>>>>> to set within the system somehow. >>>>>> What's the best way to handle it for an outgoing only system. >>>>>> I've been trying to figure out how how to configure 'Rate limiting >>>>>> concurrent outgoing calls via a provider' which is mentioned in the >>>>>> wiki a bit but nothing specific on what to actually do. >>>>>> >>>>>> *Alex Ynema** *| IT Consultant >>>>>> alex at opensystems.net.au >>>>>> >>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>> Office: +61 8 9427 2500 >>>>>> Mobile: +61 404 796 894 >>>>>> >>>>>> IT Consultant for Open Systems Support >>>>>> www.opensystems.net.au >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>> >>> >> >> >> -- >> Sincerely >> >> Jay >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/ee87f60d/attachment-0001.html From msc at freeswitch.org Wed May 29 19:29:37 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 May 2013 08:29:37 -0700 Subject: [Freeswitch-users] Errer In-Reply-To: <07BF4904977CC645B485E970424193AD130923A407@localhost> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> Message-ID: Sounds like you'll need to open a Jira on this. If you can isolate which version broke it, or ideally, which commit broke it then that would be stellar. Are you familiar with git bisect? That would assist with the troubleshooting process. -MC On Wed, May 29, 2013 at 7:41 AM, wrote: > Hi,**** > > ** ** > > I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X > bytes)" with FS 1.2.10 with t38_gateway. **** > > Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... > **** > > ** ** > > FS 1.2.5.3 works!**** > > ** ** > > Dialplan is**** > > ** ** > > **** > > **** > > **** > > **** > > ** > ** > > **** > > **** > > **** > > data="{sip_execute_on_image='t38_gateway self nocng'}sofia/intern/${ > num}@192.168.1.1:41000"/>**** > > **** > > **** > > **** > > ** ** > > Anything I can try?!**** > > ** ** > > VG,**** > > Benjamin T.**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/a3e51752/attachment.html From xiaofengcanyuexp at 163.com Wed May 29 19:46:59 2013 From: xiaofengcanyuexp at 163.com (xiaofengcanyuexp at 163.com) Date: Wed, 29 May 2013 23:46:59 +0800 Subject: [Freeswitch-users] How to get the raw ISUP from sangoma/wanpipe References: , , , , <51A5A5AD.1010605@gmail.com>, , Message-ID: <201305292346573124992@163.com> I'm taking use of below call model: SIP<--->mod_sofia ---- mod_freetdm---[sangoma]--wanpipe<-->E1 Now, I want to get the raw ISUP at [sangoma] part. The current situation is, the wanpipe provides some ISUP message structures in "sit.x", which is the extractor of raw ISUP. Anyone know how to get the raw ISUP excepte to re-encode from the structure? Is there any API? Thanks Windy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/692a4c9c/attachment.html From msc at freeswitch.org Wed May 29 20:05:54 2013 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 May 2013 09:05:54 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29 We have Martin from VoIPMonitor.org who will be sharing with us some information about the project, both the OSS piece and the commercial GUI front-end. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/1ae749a6/attachment.html From brian at freeswitch.org Wed May 29 21:00:44 2013 From: brian at freeswitch.org (Brian West) Date: Wed, 29 May 2013 12:00:44 -0500 Subject: [Freeswitch-users] 888 Today Message-ID: <17CDDC5A-FAAD-40D3-93E3-AE3D5F75106E@freeswitch.org> If you have a video enabled endpoint you can point at sip:888 at mcu.freeswitch.org with video as we have the MCU bridged to 888. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest From drk at drkngs.net Wed May 29 21:15:21 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 29 May 2013 10:15:21 -0700 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: <51A5AC2D.3040406@gmail.com> Message-ID: <20130529171521.148a5ddf@mail.tritonwest.net> A little known trick for using limit, is that you can do it on the B-LEG, rather then calling it form the A-Leg and you won't have to worry about holding the locks if the B-Leg fails, and you fall over to somethng else. If you do the limit using "execute_on_originate" and the B-LEG fails somewhere along the way, the lock will go away when the B-LEG goes away. Also if the limit fails, and returns the condition linke user_busy or something else, it will just look like the gateway failed, with that response. This is a much better way of using limit for some type of outbound applicaiton. --Dave _____ From: Muhammad Naseer Bhatti [mailto:nbhatti at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 29 May 2013 00:20:13 -0700 Subject: Re: [Freeswitch-users] Gateway Call Limits Hmm, so simple. I wonder how I overlooked it :) -- Thanks, Muhammad Naseer Bhatti jay binks wrote: http://wiki.freeswitch.org/wiki/Limit On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: Sorry for the thread hijack, but on the other hand, is it possible to limit the number of outgoing CPS? Don't seem to see that either in the wiki or a way to make it work. -- Thanks, Muhammad Naseer Bhatti Alex Ynema wrote: Cheers Avi I've now changed that to hash as I don't need it to be persistent. What should I see in the clie to confirm this is working without attempting 150+ calls Basically I've added this to my default.xml Alex Ynema | IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 14:13, Avi Marcus wrote: ... just note that's stored in a database (db) not ram (hash) so if you don't need to share it / have persistence, just store it in ram. -Avi On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: I've implemented this in default.xml hoping to limit each of my two gateways to 150. Based on what's in http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so hopefully that works. Alex Ynema | IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 13:59, Alex Ynema wrote: Hi, I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. What's the best way to handle it for an outgoing only system. I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. Alex Ynema | IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130529/6284a4ba/attachment-0001.html From eoaddai at gmail.com Wed May 29 21:23:32 2013 From: eoaddai at gmail.com (Emmanuel Owusu Addai) Date: Wed, 29 May 2013 17:23:32 +0000 Subject: [Freeswitch-users] SIP/2.0 606 Not Acceptable and INCOMPATIBLE_DESTINATION In-Reply-To: References: <1369468901841-7591099.post@n2.nabble.com> Message-ID: Hei, I've got it sorted out. I only changed the PC Port settings under the network configuration of the goip from Bridge mode to router mode. and it started working. thanks for suggestions from all of you. On 29 May 2013 12:41, eoaddai wrote: > HI, > i actually changed the goip8 to goip4 gateway, and the samme error SIP/2.0 > 606 Not Acceptable and INCOMPATIBLE_DESTINATION is been generated. Anyone > with any ideas too? > > > On 27 May 2013 19:01, Steven Ayre [via freeswitch-users] <[hidden email] > > wrote: > >> recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >>> ------------------------------------------------------------ >>> ------------ >>> SIP/2.0 606 Not Acceptable >> >> >> Yes. The 606 packet is generated by the goip and sent by the goip to >> freeswitch. >> >> We/FreeSWITCH cannot tell you why the goip sent 606, you would need to >> check its logs/records or with the provider/manufacturer to see why. >> >> The only thing I would say is a codec mismatch is normally 488 not 606, >> so chances are it's something else... but that is assuming that they are >> following the standards, which is not necessarily the case. >> >> One thing I would try is setting the verbose_sdp variable: >> >> See http://wiki.freeswitch.org/wiki/Variable_verbose_sdp for >> >> >> -Steve >> >> >> >> >> On 26 May 2013 12:20, eoaddai <[hidden email] >> > wrote: >> >>> So, is the call hitting the goip at all? or, freeswitch is unable to >>> process it to the goip? >>> >>> >>> On 26 May 2013 10:32, Daniel Ivanov [via freeswitch-users] <[hidden >>> email] > wrote: >>> >>>> Maybe you're not sending them the right codecs or trying to run a >>>> feature they don't have. Revisit your vars.xml and sip_profiles .xml. . >>>> Ultimately contact the provider to ask them what youre doing wrong in your >>>> sdps. >>>> On May 26, 2013 11:26 AM, "eoaddai" <[hidden email]> >>>> wrote: >>>> >>>>> Hi, my first time of posting stuff here. I really need help. >>>>> I keep on not getting my calls go through freeswitch to my goip >>>>> gateway. The >>>>> following is the freeswitch log with siptrace turned on i get. Please >>>>> help >>>>> me: >>>>> >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:2050 Parsing >>>>> global variables >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_request_uuid]=[fbcbf638-c50f-11e2-92cc-0050dab86386] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/1] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [origination_caller_id_number]=[0264370536] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_from]=[0264370536] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_to]=[0267577771] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_app]=[true] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [originate_timeout]=[60] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [ignore_early_media]=[true] >>>>> 2013-05-25 07:52:01.430139 [NOTICE] switch_channel.c:978 New Channel >>>>> sofia/external/0267577771 [fbce2b24-c50f-11e2-ada4-0fb75ece6ad1] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:4420 >>>>> (sofia/external/0267577771) State Change CS_NEW -> CS_INIT >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0267577771) Running State Change CS_INIT >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>>>> (sofia/external/0267577771) State INIT >>>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:87 >>>>> sofia/external/0267577771 >>>>> SOFIA INIT >>>>> 2013-05-25 07:52:01.430139 [DEBUG] sofia_glue.c:1219 Local SDP: >>>>> v=0 >>>>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.50.1 >>>>> t=0 0 >>>>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=ptime:20 >>>>> a=sendrecv >>>>> >>>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:114 >>>>> (sofia/external/0267577771) State Change CS_INIT -> CS_ROUTING >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:454 >>>>> (sofia/external/0267577771) State INIT going to sleep >>>>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.434263: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> INVITE [hidden email]SIP/2.0 >>>>> >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>>> Max-Forwards: 70 >>>>> From: "" <[hidden email] >>>>> >;tag=DXjN0tK6mcc5S >>>>> To: <[hidden email] >>>>> > >>>>> >>>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 INVITE >>>>> Contact: >>>>> User-Agent: >>>>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>>>> REGISTER, REFER, NOTIFY >>>>> Supported: timer, precondition, path, replaces >>>>> Allow-Events: talk, hold, conference, refer >>>>> Content-Type: application/sdp >>>>> Content-Disposition: session >>>>> Content-Length: 201 >>>>> X-FS-Support: update_display,send_info >>>>> Remote-Party-ID: >>>>> <[hidden email] >>>>> >;party=calling;screen=yes;privacy=off >>>>> >>>>> >>>>> v=0 >>>>> o=FreeSWITCH 1369438147 1369438148 IN IP4 10.10.50.1 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.50.1 >>>>> t=0 0 >>>>> m=audio 30174 RTP/AVP 0 8 3 101 13 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=ptime:20 >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0267577771) Running State Change CS_ROUTING >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>>>> (sofia/external/0267577771) State ROUTING >>>>> 2013-05-25 07:52:01.430139 [DEBUG] mod_sofia.c:137 >>>>> sofia/external/0267577771 >>>>> SOFIA ROUTING >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_ivr_originate.c:67 >>>>> (sofia/external/0267577771) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:470 >>>>> (sofia/external/0267577771) State ROUTING going to sleep >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0267577771) Running State Change CS_CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>>>> (sofia/external/0267577771) State CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.430139 [DEBUG] switch_core_state_machine.c:489 >>>>> (sofia/external/0267577771) State CONSUME_MEDIA going to sleep >>>>> 2013-05-25 07:52:01.430139 [DEBUG] sofia.c:5745 Channel >>>>> sofia/external/0267577771 entering state [calling][0] >>>>> recv 305 bytes from udp/[10.10.50.10]:5060 at 07:52:01.472849: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 606 Not Acceptable >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>>> From: "" <[hidden email] >>>>> >;tag=DXjN0tK6mcc5S >>>>> To: <[hidden email] >>>>> >;tag=1662509363 >>>>> >>>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 INVITE >>>>> User-Agent: dble >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 314 bytes to udp/[10.10.50.10]:5060 at 07:52:01.473196: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK [hidden email]SIP/2.0 >>>>> >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKDXtjtvNyXem0N >>>>> Max-Forwards: 70 >>>>> From: "" <[hidden email] >>>>> >;tag=DXjN0tK6mcc5S >>>>> To: <[hidden email] >>>>> >;tag=1662509363 >>>>> >>>>> Call-ID: d32cdb4c-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] sofia.c:5745 Channel >>>>> sofia/external/0267577771 entering state [terminated][606] >>>>> 2013-05-25 07:52:01.470107 [NOTICE] sofia.c:6553 Hangup >>>>> sofia/external/0267577771 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_ivr_originate.c:3617 >>>>> Originate >>>>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_channel.c:3053 Send signal >>>>> sofia/external/0267577771 [KILL] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0267577771) Running State Change CS_HANGUP >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>>>> (sofia/external/0267577771) State HANGUP >>>>> 2013-05-25 07:52:01.470107 [DEBUG] mod_sofia.c:463 Channel >>>>> sofia/external/0267577771 hanging up, cause: INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:48 >>>>> sofia/external/0267577771 Standard HANGUP, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:676 >>>>> (sofia/external/0267577771) State HANGUP going to sleep >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:689 >>>>> (sofia/external/0267577771) Callstate Change DOWN -> HANGUP >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:446 >>>>> (sofia/external/0267577771) State Change CS_HANGUP -> CS_REPORTING >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0267577771) Running State Change CS_REPORTING >>>>> 2013-05-25 07:52:01.470107 [DEBUG] switch_core_state_machine.c:761 >>>>> (sofia/external/0267577771) State REPORTING >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:92 >>>>> sofia/external/0267577771 Standard REPORTING, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:761 >>>>> (sofia/external/0267577771) State REPORTING going to sleep >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:440 >>>>> (sofia/external/0267577771) State Change CS_REPORTING -> CS_DESTROY >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0267577771 [BREAK] >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_session.c:1549 Session 3 >>>>> (sofia/external/0267577771) Locked, Waiting on external entities >>>>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1567 Session >>>>> 3 >>>>> (sofia/external/0267577771) Ended >>>>> 2013-05-25 07:52:01.550124 [NOTICE] switch_core_session.c:1571 Close >>>>> Channel >>>>> sofia/external/0267577771 [CS_DESTROY] >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:565 >>>>> (sofia/external/0267577771) Callstate Change HANGUP -> DOWN >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:568 >>>>> (sofia/external/0267577771) Running State Change CS_DESTROY >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>>>> (sofia/external/0267577771) State DESTROY >>>>> 2013-05-25 07:52:01.550124 [DEBUG] mod_sofia.c:373 >>>>> sofia/external/0267577771 >>>>> SOFIA DESTROY >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:99 >>>>> sofia/external/0267577771 Standard DESTROY >>>>> 2013-05-25 07:52:01.550124 [DEBUG] switch_core_state_machine.c:578 >>>>> (sofia/external/0267577771) State DESTROY going to sleep >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_ivr_originate.c:2050 Parsing >>>>> global variables >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_request_uuid]=[fbe6ddb8-c50f-11e2-92cc-0050dab86386] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_answer_url]=[http://127.0.0.1/deliverylogs/answer/2] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_ring_url]=[http://127.0.0.1/CallQueue/ring.php] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_hangup_url]=[http://127.0.0.1/CallQueue/hangup.php] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [origination_caller_id_number]=[0264370536] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_from]=[0264370536] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_to]=[0249230704] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [plivo_app]=[true] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [originate_timeout]=[60] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_event.c:1608 Parsing variable >>>>> [ignore_early_media]=[true] >>>>> 2013-05-25 07:52:01.590098 [NOTICE] switch_channel.c:978 New Channel >>>>> sofia/external/0249230704 [fbe94224-c50f-11e2-ada8-0fb75ece6ad1] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] mod_sofia.c:4420 >>>>> (sofia/external/0249230704) State Change CS_NEW -> CS_INIT >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.590098 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0249230704) Running State Change CS_INIT >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>>>> (sofia/external/0249230704) State INIT >>>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:87 >>>>> sofia/external/0249230704 >>>>> SOFIA INIT >>>>> 2013-05-25 07:52:01.610137 [DEBUG] sofia_glue.c:1219 Local SDP: >>>>> v=0 >>>>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.50.1 >>>>> t=0 0 >>>>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=ptime:20 >>>>> a=sendrecv >>>>> >>>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:114 >>>>> (sofia/external/0249230704) State Change CS_INIT -> CS_ROUTING >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:454 >>>>> (sofia/external/0249230704) State INIT going to sleep >>>>> send 1011 bytes to udp/[10.10.50.10]:5060 at 07:52:01.612380: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> INVITE [hidden email]SIP/2.0 >>>>> >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>>> Max-Forwards: 70 >>>>> From: "" <[hidden email] >>>>> >;tag=e6Be2N49HN2QN >>>>> To: <[hidden email] >>>>> > >>>>> >>>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 INVITE >>>>> Contact: >>>>> User-Agent: >>>>> FreeSWITCH-mod_sofia/1.5.2b+git~20130525T032404Z~12f2f674f9 >>>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>>>> REGISTER, REFER, NOTIFY >>>>> Supported: timer, precondition, path, replaces >>>>> Allow-Events: talk, hold, conference, refer >>>>> Content-Type: application/sdp >>>>> Content-Disposition: session >>>>> Content-Length: 201 >>>>> X-FS-Support: update_display,send_info >>>>> Remote-Party-ID: >>>>> <[hidden email] >>>>> >;party=calling;screen=yes;privacy=off >>>>> >>>>> >>>>> v=0 >>>>> o=FreeSWITCH 1369449071 1369449072 IN IP4 10.10.50.1 >>>>> s=FreeSWITCH >>>>> c=IN IP4 10.10.50.1 >>>>> t=0 0 >>>>> m=audio 19250 RTP/AVP 0 8 3 101 13 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:101 0-16 >>>>> a=ptime:20 >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0249230704) Running State Change CS_ROUTING >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>>>> (sofia/external/0249230704) State ROUTING >>>>> 2013-05-25 07:52:01.610137 [DEBUG] mod_sofia.c:137 >>>>> sofia/external/0249230704 >>>>> SOFIA ROUTING >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_ivr_originate.c:67 >>>>> (sofia/external/0249230704) State Change CS_ROUTING -> CS_CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:470 >>>>> (sofia/external/0249230704) State ROUTING going to sleep >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0249230704) Running State Change CS_CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>>>> (sofia/external/0249230704) State CONSUME_MEDIA >>>>> 2013-05-25 07:52:01.610137 [DEBUG] switch_core_state_machine.c:489 >>>>> (sofia/external/0249230704) State CONSUME_MEDIA going to sleep >>>>> 2013-05-25 07:52:01.610137 [DEBUG] sofia.c:5745 Channel >>>>> sofia/external/0249230704 entering state [calling][0] >>>>> recv 304 bytes from udp/[10.10.50.10]:5060 at 07:52:01.633202: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> SIP/2.0 606 Not Acceptable >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>>> From: "" <[hidden email] >>>>> >;tag=e6Be2N49HN2QN >>>>> To: <[hidden email] >>>>> >;tag=633680086 >>>>> >>>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 INVITE >>>>> User-Agent: dble >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> send 313 bytes to udp/[10.10.50.10]:5060 at 07:52:01.633558: >>>>> >>>>> ------------------------------------------------------------------------ >>>>> ACK [hidden email]SIP/2.0 >>>>> >>>>> Via: SIP/2.0/UDP 10.10.50.1:5080;rport;branch=z9hG4bKe6KBvQ61tQaKH >>>>> Max-Forwards: 70 >>>>> From: "" <[hidden email] >>>>> >;tag=e6Be2N49HN2QN >>>>> To: <[hidden email] >>>>> >;tag=633680086 >>>>> >>>>> Call-ID: d3480e87-3fb2-1231-0dab-0050dab86386 >>>>> CSeq: 44398096 ACK >>>>> Content-Length: 0 >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1006 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] sofia.c:5745 Channel >>>>> sofia/external/0249230704 entering state [terminated][606] >>>>> 2013-05-25 07:52:01.630102 [NOTICE] sofia.c:6553 Hangup >>>>> sofia/external/0249230704 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_channel.c:3053 Send signal >>>>> sofia/external/0249230704 [KILL] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0249230704) Running State Change CS_HANGUP >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>>>> (sofia/external/0249230704) State HANGUP >>>>> 2013-05-25 07:52:01.630102 [DEBUG] mod_sofia.c:463 Channel >>>>> sofia/external/0249230704 hanging up, cause: INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:48 >>>>> sofia/external/0249230704 Standard HANGUP, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:676 >>>>> (sofia/external/0249230704) State HANGUP going to sleep >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:689 >>>>> (sofia/external/0249230704) Callstate Change DOWN -> HANGUP >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:446 >>>>> (sofia/external/0249230704) State Change CS_HANGUP -> CS_REPORTING >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:415 >>>>> (sofia/external/0249230704) Running State Change CS_REPORTING >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>>>> (sofia/external/0249230704) State REPORTING >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:92 >>>>> sofia/external/0249230704 Standard REPORTING, cause: >>>>> INCOMPATIBLE_DESTINATION >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:761 >>>>> (sofia/external/0249230704) State REPORTING going to sleep >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_state_machine.c:440 >>>>> (sofia/external/0249230704) State Change CS_REPORTING -> CS_DESTROY >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1341 Send >>>>> signal >>>>> sofia/external/0249230704 [BREAK] >>>>> 2013-05-25 07:52:01.630102 [DEBUG] switch_core_session.c:1549 Session 4 >>>>> (sofia/external/0249230704) Locked, Waiting on external entities >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_ivr_originate.c:3617 >>>>> Originate >>>>> Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] >>>>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1567 Session >>>>> 4 >>>>> (sofia/external/0249230704) Ended >>>>> 2013-05-25 07:52:01.650227 [NOTICE] switch_core_session.c:1571 Close >>>>> Channel >>>>> sofia/external/0249230704 [CS_DESTROY] >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:565 >>>>> (sofia/external/0249230704) Callstate Change HANGUP -> DOWN >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:568 >>>>> (sofia/external/0249230704) Running State Change CS_DESTROY >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>>>> (sofia/external/0249230704) State DESTROY >>>>> 2013-05-25 07:52:01.650227 [DEBUG] mod_sofia.c:373 >>>>> sofia/external/0249230704 >>>>> SOFIA DESTROY >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:99 >>>>> sofia/external/0249230704 Standard DESTROY >>>>> 2013-05-25 07:52:01.650227 [DEBUG] switch_core_state_machine.c:578 >>>>> (sofia/external/0249230704) State DESTROY going to sleep >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> View this message in context: >>>>> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099.html >>>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> [hidden email] >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> [hidden email] >>>>> >>>>> 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: >>>> [hidden email] >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> [hidden email] >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ------------------------------ >>>> If you reply to this email, your message will be added to the >>>> discussion below: >>>> >>>> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591120.html >>>> To unsubscribe from SIP/2.0 606 Not Acceptable and >>>> INCOMPATIBLE_DESTINATION, click here. >>>> NAML >>>> >>> >>> >>> >>> -- >>> *Emmanuel O. Addai,* >>> *>> target="_blank">+233(0)26 757 7771* >>> >>> ------------------------------ >>> View this message in context: Re: SIP/2.0 606 Not Acceptable and >>> INCOMPATIBLE_DESTINATION >>> >>> Sent from the freeswitch-users mailing list archiveat Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> [hidden email] >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> [hidden email] >>> 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: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> If you reply to this email, your message will be added to the >> discussion below: >> >> http://freeswitch-users.2379917.n2.nabble.com/SIP-2-0-606-Not-Acceptable-and-INCOMPATIBLE-DESTINATION-tp7591099p7591155.html >> To unsubscribe from SIP/2.0 606 Not Acceptable and >> INCOMPATIBLE_DESTINATION, click here. >> NAML >> > > > > -- > *Emmanuel O. Addai,* > *+233(0)26 757 7771* > > ------------------------------ > View this message in context: Re: SIP/2.0 606 Not Acceptable and > INCOMPATIBLE_DESTINATION > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- *Emmanuel O. Addai,* *+233(0)26 757 7771* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/dcccbc13/attachment-0001.html From anthony.minessale at gmail.com Wed May 29 22:19:02 2013 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 29 May 2013 13:19:02 -0500 Subject: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE In-Reply-To: References: Message-ID: Hangup is when it hangs up, hangup_complete is after the cdr processing is complete right before destroy. On Tue, May 28, 2013 at 3:50 PM, Tihomir Culjaga wrote: > hello > > > im wondering what should we track as hangup event ... CHANNEL_HANGUP or > CHANNEL_HANGUP_COMPLETE ... > > i have a situation where A calls FS, an ESL application answers this call, > originates a new call to B and bridges A and B... than ESL originate > another call towards C and joins all 3 channels into a conference. > > now , A hangs up, FS sends just CHANNEL_HANGUP event ... not > CHANNEL_HANGUP_COMPLETE ... > > > im wondering if it is supposed to be like that or we need to get > CHANNEL_HANGUP_COMPLETE ? > > > T. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/345ec773/attachment.html From drk at drkngs.net Wed May 29 23:02:29 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 29 May 2013 12:02:29 -0700 Subject: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE In-Reply-To: Message-ID: <20130529190229.38fe4428@mail.tritonwest.net> Also worth noting, CHANNEL_HANGUP_COMPLETE is the event you use to write CDRs if you're going to do your own CDR module, or do it external. --Dave _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 29 May 2013 11:19:02 -0700 Subject: Re: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE Hangup is when it hangs up, hangup_complete is after the cdr processing is complete right before destroy. On Tue, May 28, 2013 at 3:50 PM, Tihomir Culjaga wrote: hello im wondering what should we track as hangup event ... CHANNEL_HANGUP or CHANNEL_HANGUP_COMPLETE ... i have a situation where A calls FS, an ESL application answers this call, originates a new call to B and bridges A and B... than ESL originate another call towards C and joins all 3 channels into a conference. now , A hangs up, FS sends just CHANNEL_HANGUP event ... not CHANNEL_HANGUP_COMPLETE ... im wondering if it is supposed to be like that or we need to get CHANNEL_HANGUP_COMPLETE ? T. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/8e02aa15/attachment.html From mike at jerris.com Wed May 29 23:56:43 2013 From: mike at jerris.com (Michael Jerris) Date: Wed, 29 May 2013 15:56:43 -0400 Subject: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE In-Reply-To: <20130529190229.38fe4428@mail.tritonwest.net> References: <20130529190229.38fe4428@mail.tritonwest.net> Message-ID: <699859B0-6E1D-49BF-AA48-F808D9263E4C@jerris.com> If you are writing a cdr module, you should be using the on_reporting state handler, not the event handler for hangup complete. Mike On May 29, 2013, at 3:02 PM, Dave R. Kompel wrote: > Also worth noting, CHANNEL_HANGUP_COMPLETE is the event you use to write CDRs if you're going to do your own CDR module, or do it external. > > --Dave > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Wed, 29 May 2013 11:19:02 -0700 > Subject: Re: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE > > Hangup is when it hangs up, hangup_complete is after the cdr processing is complete right before destroy. > > > > On Tue, May 28, 2013 at 3:50 PM, Tihomir Culjaga wrote: > hello > > > im wondering what should we track as hangup event ... CHANNEL_HANGUP or CHANNEL_HANGUP_COMPLETE ... > > i have a situation where A calls FS, an ESL application answers this call, originates a new call to B and bridges A and B... than ESL originate another call towards C and joins all 3 channels into a conference. > > now , A hangs up, FS sends just CHANNEL_HANGUP event ... not CHANNEL_HANGUP_COMPLETE ... > > > im wondering if it is supposed to be like that or we need to get CHANNEL_HANGUP_COMPLETE ? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/a4b7adeb/attachment.html From red.rain.seven at gmail.com Thu May 30 00:11:27 2013 From: red.rain.seven at gmail.com (Henry Huang) Date: Wed, 29 May 2013 13:11:27 -0700 Subject: [Freeswitch-users] 888 Today In-Reply-To: <17CDDC5A-FAAD-40D3-93E3-AE3D5F75106E@freeswitch.org> References: <17CDDC5A-FAAD-40D3-93E3-AE3D5F75106E@freeswitch.org> Message-ID: Is this a external MCU or build-in to FreeSWITCH? Henry On Wed, May 29, 2013 at 10:00 AM, Brian West wrote: > If you have a video enabled endpoint you can point at > sip:888 at mcu.freeswitch.org with video as we have the MCU bridged to 888. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > 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://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/20130529/6da8cca9/attachment-0001.html From anton.vazir at gmail.com Thu May 30 00:23:59 2013 From: anton.vazir at gmail.com (Anton VG) Date: Thu, 30 May 2013 00:23:59 +0400 Subject: [Freeswitch-users] no ESL event for SIP SIMPLE message if destination number prefixed by + Message-ID: Hi! Just was playing with SIP SIMPLE ESL messages, actually all works fine if message sent to 1234567890 at freeswitch.host. BUT if message is sent to +1234567890 at freeswitch.host (+ prefix!)- no event fired :( I can see the PCAP the message accepted by FS the same way. So + somehow forces fs to not fire the message. Did anyone encountered such behavior, so should I fill the bug report on JIRA? Regards, Anton -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/8dc91373/attachment.html From peter at olssononline.se Thu May 30 01:11:46 2013 From: peter at olssononline.se (Peter Olsson) Date: Wed, 29 May 2013 23:11:46 +0200 Subject: [Freeswitch-users] no ESL event for SIP SIMPLE message if destination number prefixed by + In-Reply-To: References: Message-ID: Sounds like a bug to me - so please go ahead and file a JIra. 2013/5/29 Anton VG > Hi! > > Just was playing with SIP SIMPLE ESL messages, actually all works fine if > message sent to 1234567890 at freeswitch.host. > BUT if message is sent to +1234567890 at freeswitch.host (+ prefix!)- no > event fired :( I can see the PCAP the message accepted by FS the same way. > So + somehow forces fs to not fire the message. Did anyone encountered such > behavior, so should I fill the bug report on JIRA? > > Regards, > Anton > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/70233470/attachment.html From krice at freeswitch.org Thu May 30 01:18:08 2013 From: krice at freeswitch.org (Ken Rice) Date: Wed, 29 May 2013 16:18:08 -0500 Subject: [Freeswitch-users] 888 Today In-Reply-To: Message-ID: Its an external MCU On 5/29/13 3:11 PM, "Henry Huang" wrote: > Is this a external MCU or build-in to FreeSWITCH? > > Henry > > > On Wed, May 29, 2013 at 10:00 AM, Brian West wrote: >> If you have a video enabled endpoint you can point at >> sip:888 at mcu.freeswitch.org with video >> as we have the MCU bridged to 888. >> -- >> Brian West >> brian at freeswitch.org >> FreeSWITCH Solutions, LLC >> PO BOX PO BOX 2531 >> Brookfield, WI 53008-2531 >> Twitter: @FreeSWITCH_Wire >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 ?| ?F: +1.918.420.9002 >> ?| ?M: +1.918.424.WEST >> 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://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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/76c6a7ca/attachment.html From william.king at quentustech.com Thu May 30 01:26:52 2013 From: william.king at quentustech.com (William King) Date: Wed, 29 May 2013 14:26:52 -0700 Subject: [Freeswitch-users] no ESL event for SIP SIMPLE message if destination number prefixed by + In-Reply-To: References: Message-ID: <51A6729C.1020404@quentustech.com> Does your chat plan attempt to match with the + prefix? 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 05/29/2013 01:23 PM, Anton VG wrote: > Hi! > > Just was playing with SIP SIMPLE ESL messages, actually all works fine > if message sent to 1234567890 at freeswitch.host. > BUT if message is sent to +1234567890 at freeswitch.host (+ prefix!)- no > event fired :( I can see the PCAP the message accepted by FS the same > way. So + somehow forces fs to not fire the message. Did anyone > encountered such behavior, so should I fill the bug report on JIRA? > > Regards, > Anton > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From tru083 at yahoo.com Thu May 30 03:13:50 2013 From: tru083 at yahoo.com (D D) Date: Wed, 29 May 2013 16:13:50 -0700 (PDT) Subject: [Freeswitch-users] How can I cancel a play_and_get_digits? Message-ID: <1369869230.89941.YahooMailNeo@web120702.mail.ne1.yahoo.com> Hi, I am using an play_and_get_digits in an ESL program. ? Based on an external event, I need to cancel the?play_and_get_digits operation. If I send a "break", it only seems to stop the audio from playing, but the? play_and_get_digits operation continues, and later the response event comes. How can I stop the entire play_and_get_digits operation from ESL? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/d2e1f4b0/attachment.html From drk at drkngs.net Thu May 30 03:41:34 2013 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 29 May 2013 16:41:34 -0700 Subject: [Freeswitch-users] =?iso-8859-1?q?How_can_I_cancel_a_play=5Fand?= =?iso-8859-1?q?=5Fget=5Fdigits=3F?= In-Reply-To: <1369869230.89941.YahooMailNeo@web120702.mail.ne1.yahoo.com> Message-ID: <20130529234134.9d4e7dd4@mail.tritonwest.net> You can't, at least from what I can tell. Look at the source code for it in mod_dptools. If you need this control over it you may want to implment DTMF buffers in your ESL app, from DTMF events, and do it all your self. The othe option you have is to write your own DialPaln app (scripting language or something) that can comunicate out of band via events to your ESL program, that you use for DTMF input. What is your application? Maybe ESL isn't the right tool for the job. It seems like for some reason, everyone jumps to ESL as the only recomended and supported way of doing things external. It is NOT. The new book that should be available does a better job of clearing that up. Can't wait for the new book to be released!!!! --Dave _____ From: D D [mailto:tru083 at yahoo.com] To: freeswitch-users at lists.freeswitch.org [mailto:freeswitch-users at lists.freeswitch.org] Sent: Wed, 29 May 2013 16:13:50 -0700 Subject: [Freeswitch-users] How can I cancel a play_and_get_digits? Hi, I am using an play_and_get_digits in an ESL program. Based on an external event, I need to cancel the play_and_get_digits operation. If I send a "break", it only seems to stop the audio from playing, but the play_and_get_digits operation continues, and later the response event comes. How can I stop the entire play_and_get_digits operation from ESL? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130529/01360f3d/attachment-0001.html From grcamauer at gmail.com Thu May 30 05:38:19 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 29 May 2013 22:38:19 -0300 Subject: [Freeswitch-users] How can I cancel a play_and_get_digits? In-Reply-To: <20130529234134.9d4e7dd4@mail.tritonwest.net> References: <20130529234134.9d4e7dd4@mail.tritonwest.net> Message-ID: <2745495820513918199@unknownmsgid> I already bought mine and read several chapters! I particularly liked the one on security. Guillermo Sent from my iPhone On 29/05/2013, at 20:44, "Dave R. Kompel" wrote: You can't, at least from what I can tell. Look at the source code for it in mod_dptools. If you need this control over it you may want to implment DTMF buffers in your ESL app, from DTMF events, and do it all your self. The othe option you have is to write your own DialPaln app (scripting language or something) that can comunicate out of band via events to your ESL program, that you use for DTMF input. What is your application? Maybe ESL isn't the right tool for the job. It seems like for some reason, everyone jumps to ESL as the only recomended and supported way of doing things external. It is NOT. The new book that should be available does a better job of clearing that up. Can't wait for the new book to be released!!!! --Dave ------------------------------ *From:* D D [mailto:tru083 at yahoo.com ] *To:* freeswitch-users at lists.freeswitch.org [ mailto:freeswitch-users at lists.freeswitch.org ] *Sent:* Wed, 29 May 2013 16:13:50 -0700 *Subject:* [Freeswitch-users] How can I cancel a play_and_get_digits? Hi, I am using an play_and_get_digits in an ESL program. Based on an external event, I need to cancel the play_and_get_digits operation. If I send a "break", it only seems to stop the audio from playing, but the play_and_get_digits operation continues, and later the response event comes. How can I stop the entire play_and_get_digits operation from ESL? Thanks, David _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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/20130529/bb8749b5/attachment.html From luis.daniel.lucio at gmail.com Thu May 30 05:42:36 2013 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 29 May 2013 21:42:36 -0400 Subject: [Freeswitch-users] OT: sonetel incopatible with fusionbpx/freeswitch??? Message-ID: DOes anybody has sucessfully use sonetel to do SIP forward to another port different than 5060. It seems that what ever string configuration: mydomain:5080 number at mydomain:5080 sip:number at mydomain:5080 the To field is sent okay with correct port, but TCPdump capture shows me that they are hitting port 5060/udp Just wondering if someone has sucessfully configure it From bdfoster at davri.com Thu May 30 05:58:38 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 21:58:38 -0400 Subject: [Freeswitch-users] OT: sonetel incopatible with fusionbpx/freeswitch??? In-Reply-To: References: Message-ID: Sonus gateways are known to operate in non-standard or weird ways, so it doesn't surprise me that you're running into this. But for safety's sake, please post a console log of a call where this happens with siptrace of external profile ('sofia profile external siptrace on' without quotes and performed in fs_cli) to pastebin.freeswitch.org. - BDF On May 29, 2013 9:48 PM, "Luis Daniel Lucio Quiroz" < luis.daniel.lucio at gmail.com> wrote: > DOes anybody has sucessfully use sonetel to do SIP forward to another > port different than 5060. It seems that what ever string > configuration: > > mydomain:5080 > number at mydomain:5080 > sip:number at mydomain:5080 > > the To field is sent okay with correct port, but TCPdump capture shows > me that they are hitting port 5060/udp > > Just wondering if someone has sucessfully configure it > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/513c8bca/attachment.html From alex at opensystems.net.au Thu May 30 06:15:03 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Thu, 30 May 2013 10:15:03 +0800 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: What do you mean by ESL Program. This system is running a newfies-dialer frontend with a pretty stock install of Freeswitch. *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 29 May 2013 23:22, Michael Collins wrote: > It sounds like you have an ESL program that is not consuming events > quickly enough and the event queue is filling up. > -MC > > > On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: > >> so setting the limit at 150 was fine but as soon as I set it to 200 I've >> now hit a problem. >> Freeswitch has slowly grown to 372 sessions and getting lots of these >> errors in the cli >> >> 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more >> events..... >> >> UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, >> 606 microseconds >> FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >> 1500 session(s) since startup >> 372 session(s) - 0 out of max 10 per sec >> 10000 session(s) max >> min idle cpu 0.00/100.00 >> Current Stack Size/Max 240K/8192K >> >> >> >> *Alex Ynema** *| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 15:01, jay binks wrote: >> >>> http://wiki.freeswitch.org/wiki/Limit >>> >>> >>> >>> >>> On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >>> >>>> >>>> Sorry for the thread hijack, but on the other hand, is it possible to >>>> limit the number of outgoing CPS? Don't seem to see that either in the wiki >>>> or a way to make it work. >>>> >>>> -- >>>> Thanks, >>>> Muhammad Naseer Bhatti >>>> >>>> >>>> >>>> Alex Ynema wrote: >>>> >>>> Cheers Avi I've now changed that to hash as I don't need it to be >>>> persistent. >>>> What should I see in the clie to confirm this is working without >>>> attempting 150+ calls >>>> >>>> Basically I've added this to my default.xml >>>> >>>> >>>> >>> data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>>> >>>> >>>> >>> expression="zetta-cisco-1"> >>>> >>>> >>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>> >>>> >>>> >>>> >>> expression="zetta-cisco-2"> >>>> >>>> >>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>> >>>> >>>> >>>> >>>> *Alex Ynema** *| IT Consultant >>>> alex at opensystems.net.au >>>> >>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>> Office: +61 8 9427 2500 >>>> Mobile: +61 404 796 894 >>>> >>>> IT Consultant for Open Systems Support >>>> www.opensystems.net.au >>>> >>>> >>>> On 29 May 2013 14:13, Avi Marcus wrote: >>>> >>>>> ... just note that's stored in a database (db) not ram (hash) so if >>>>> you don't need to share it / have persistence, just store it in ram. >>>>> >>>>> -Avi >>>>> >>>>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >>>>> >>>>>> I've implemented this in default.xml hoping to limit each of my two >>>>>> gateways to 150. >>>>>> Based on what's in >>>>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>>>> hopefully that works. >>>>>> >>>>>> >>>>>> >>>>> expression="zetta-cisco-1"> >>>>>> >>>>>> >>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="zetta-cisco-2"> >>>>>> >>>>>> >>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> *Alex Ynema** *| IT Consultant >>>>>> alex at opensystems.net.au >>>>>> >>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>> Office: +61 8 9427 2500 >>>>>> Mobile: +61 404 796 894 >>>>>> >>>>>> IT Consultant for Open Systems Support >>>>>> www.opensystems.net.au >>>>>> >>>>>> >>>>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>>>> >>>>>>> Hi, >>>>>>> I'm trying to figure out how to limit the number of calls a Gateway >>>>>>> is allowed to use. Our Sip provider has provided up with 200 which I need >>>>>>> to set within the system somehow. >>>>>>> What's the best way to handle it for an outgoing only system. >>>>>>> I've been trying to figure out how how to configure 'Rate limiting >>>>>>> concurrent outgoing calls via a provider' which is mentioned in the >>>>>>> wiki a bit but nothing specific on what to actually do. >>>>>>> >>>>>>> *Alex Ynema** *| IT Consultant >>>>>>> alex at opensystems.net.au >>>>>>> >>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>> Office: +61 8 9427 2500 >>>>>>> Mobile: +61 404 796 894 >>>>>>> >>>>>>> IT Consultant for Open Systems Support >>>>>>> www.opensystems.net.au >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>> >>>> >>> >>> >>> -- >>> Sincerely >>> >>> Jay >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130530/b4b9f188/attachment-0001.html From bdfoster at davri.com Thu May 30 06:25:06 2013 From: bdfoster at davri.com (Brian Foster) Date: Wed, 29 May 2013 22:25:06 -0400 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: Newfies Dialer uses the ESL. See http://www.newfies-dialer.org/documentation/how-it-works/ - BDF On May 29, 2013 10:22 PM, "Alex Ynema" wrote: > What do you mean by ESL Program. This system is running a newfies-dialer > frontend with a pretty stock install of Freeswitch. > > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 29 May 2013 23:22, Michael Collins wrote: > >> It sounds like you have an ESL program that is not consuming events >> quickly enough and the event queue is filling up. >> -MC >> >> >> On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: >> >>> so setting the limit at 150 was fine but as soon as I set it to 200 I've >>> now hit a problem. >>> Freeswitch has slowly grown to 372 sessions and getting lots of these >>> errors in the cli >>> >>> 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more >>> events..... >>> >>> UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, >>> 606 microseconds >>> FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >>> 1500 session(s) since startup >>> 372 session(s) - 0 out of max 10 per sec >>> 10000 session(s) max >>> min idle cpu 0.00/100.00 >>> Current Stack Size/Max 240K/8192K >>> >>> >>> >>> *Alex Ynema** *| IT Consultant >>> alex at opensystems.net.au >>> >>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>> Office: +61 8 9427 2500 >>> Mobile: +61 404 796 894 >>> >>> IT Consultant for Open Systems Support >>> www.opensystems.net.au >>> >>> >>> On 29 May 2013 15:01, jay binks wrote: >>> >>>> http://wiki.freeswitch.org/wiki/Limit >>>> >>>> >>>> >>>> >>>> On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >>>> >>>>> >>>>> Sorry for the thread hijack, but on the other hand, is it possible to >>>>> limit the number of outgoing CPS? Don't seem to see that either in the wiki >>>>> or a way to make it work. >>>>> >>>>> -- >>>>> Thanks, >>>>> Muhammad Naseer Bhatti >>>>> >>>>> >>>>> >>>>> Alex Ynema wrote: >>>>> >>>>> Cheers Avi I've now changed that to hash as I don't need it to be >>>>> persistent. >>>>> What should I see in the clie to confirm this is working without >>>>> attempting 150+ calls >>>>> >>>>> Basically I've added this to my default.xml >>>>> >>>>> >>>>> >>>> data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>>>> >>>>> >>>>> >>>> expression="zetta-cisco-1"> >>>>> >>>>> >>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>> >>>>> >>>>> >>>>> >>>> expression="zetta-cisco-2"> >>>>> >>>>> >>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>> >>>>> >>>>> >>>>> >>>>> *Alex Ynema** *| IT Consultant >>>>> alex at opensystems.net.au >>>>> >>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>> Office: +61 8 9427 2500 >>>>> Mobile: +61 404 796 894 >>>>> >>>>> IT Consultant for Open Systems Support >>>>> www.opensystems.net.au >>>>> >>>>> >>>>> On 29 May 2013 14:13, Avi Marcus wrote: >>>>> >>>>>> ... just note that's stored in a database (db) not ram (hash) so if >>>>>> you don't need to share it / have persistence, just store it in ram. >>>>>> >>>>>> -Avi >>>>>> >>>>>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >>>>>> >>>>>>> I've implemented this in default.xml hoping to limit each of my two >>>>>>> gateways to 150. >>>>>>> Based on what's in >>>>>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>>>>> hopefully that works. >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="zetta-cisco-1"> >>>>>>> >>>>>>> >>>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="zetta-cisco-2"> >>>>>>> >>>>>>> >>>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> *Alex Ynema** *| IT Consultant >>>>>>> alex at opensystems.net.au >>>>>>> >>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>> Office: +61 8 9427 2500 >>>>>>> Mobile: +61 404 796 894 >>>>>>> >>>>>>> IT Consultant for Open Systems Support >>>>>>> www.opensystems.net.au >>>>>>> >>>>>>> >>>>>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> I'm trying to figure out how to limit the number of calls a Gateway >>>>>>>> is allowed to use. Our Sip provider has provided up with 200 which I need >>>>>>>> to set within the system somehow. >>>>>>>> What's the best way to handle it for an outgoing only system. >>>>>>>> I've been trying to figure out how how to configure 'Rate limiting >>>>>>>> concurrent outgoing calls via a provider' which is mentioned in >>>>>>>> the wiki a bit but nothing specific on what to actually do. >>>>>>>> >>>>>>>> *Alex Ynema** *| IT Consultant >>>>>>>> alex at opensystems.net.au >>>>>>>> >>>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>> Office: +61 8 9427 2500 >>>>>>>> Mobile: +61 404 796 894 >>>>>>>> >>>>>>>> IT Consultant for Open Systems Support >>>>>>>> www.opensystems.net.au >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Sincerely >>>> >>>> Jay >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130529/2f277ade/attachment-0001.html From gabe at gundy.org Thu May 30 08:52:37 2013 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 29 May 2013 22:52:37 -0600 Subject: [Freeswitch-users] DTMF outbound call In-Reply-To: References: <008101ce51af$58c3d9c0$0a4b8d40$@bluetel.fr> <7a81a430-c54a-4ccb-9369-86167436744d@email.android.com> <193B089F-C051-4A40-8980-EECD28BF69E6@freeswitch.org> <540c84ce83a8a573d67e9b7cabda8435@bluetel.fr> <6d476bf40ad663df9f42f2edc1af0e46@bluetel.fr> <81944c1e-23e5-43c3-bb58-3ca128a946f8@email.android.com> <1118828E-93E0-4832-A45C-D35ADCB05DEF@jerris.com> <990BB742-FDCE-4B43-A2BB-1585FDB735AC@jerris.com> <92E30BD5-0416-46F8-A1C8-5A912826E24E@jerris.com> <9e397857971309b1cf47340345721e94@bluetel.fr> <2c381f8aeab58684f6bb4418c469f0a0@bluetel.fr> <529AA682-486C-4760-B25D-3CE904E82109@jerris.com> <6d1224a2bc82b50a0eb9e2325faad748@bluetel.fr> <80CE128A-426D-4E27-BD5E-8DE7E85B204C@jerris.com> <0a4c01ce5bd3$fedd8300$fc988900$@bluetel.fr> <0a6001ce5bda$895d3bf0$9c17b3d0$@bluetel.fr> <0a9501ce5be1$5b97d750$12c785f0$@bluetel.fr> <0ab001ce5be5$3a5c29c0$af147d40$@bluetel.fr> Message-ID: On Wed, May 29, 2013 at 1:04 AM, wrote: > it was firewall problem. now i have the audio, but i don't see dtmf > again. on wireshark i don't see too. > http://bluetelconnect.fr/tcpdumpdtmf3.log > > tks What ever happened to trimming emails? I just trimmed nearly 500 lines when replying here. If you don't take care of the mailing list, it has a harder time taking care of you. Best, Gabe From anton.vazir at gmail.com Thu May 30 09:41:58 2013 From: anton.vazir at gmail.com (Anton VG) Date: Thu, 30 May 2013 09:41:58 +0400 Subject: [Freeswitch-users] no ESL event for SIP SIMPLE message if destination number prefixed by + In-Reply-To: <51A6729C.1020404@quentustech.com> References: <51A6729C.1020404@quentustech.com> Message-ID: I left an empty extension in chat plan, matching (.*). But I do not relay on chat plan, I connect ESL INBOUND or do /events plain all in fs_cli and send messages the same way. There is just no any events if i prefix the number with "+". Regardless, I have tried with \+?(.*) condition in chat plan and synthetic reply. Does not work anyway. 2013/5/30 William King > Does your chat plan attempt to match with the + prefix? > > 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 05/29/2013 01:23 PM, Anton VG wrote: > > Hi! > > > > Just was playing with SIP SIMPLE ESL messages, actually all works fine > > if message sent to 1234567890 at freeswitch.host. > > BUT if message is sent to +1234567890 at freeswitch.host (+ prefix!)- no > > event fired :( I can see the PCAP the message accepted by FS the same > > way. So + somehow forces fs to not fire the message. Did anyone > > encountered such behavior, so should I fill the bug report on JIRA? > > > > Regards, > > Anton > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130530/a330aeda/attachment.html From alex at opensystems.net.au Thu May 30 10:00:47 2013 From: alex at opensystems.net.au (Alex Ynema) Date: Thu, 30 May 2013 14:00:47 +0800 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: Okay so it's a fault with newfies rather than a problem with Freeswitch. Any suggestions on where I should look to work out what exactly is happening/not happening with those Event Sessions not being cleared out from the process. *Alex Ynema** *| IT Consultant alex at opensystems.net.au Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 Office: +61 8 9427 2500 Mobile: +61 404 796 894 IT Consultant for Open Systems Support www.opensystems.net.au On 30 May 2013 10:25, Brian Foster wrote: > Newfies Dialer uses the ESL. See > http://www.newfies-dialer.org/documentation/how-it-works/ > > - BDF > On May 29, 2013 10:22 PM, "Alex Ynema" wrote: > >> What do you mean by ESL Program. This system is running a newfies-dialer >> frontend with a pretty stock install of Freeswitch. >> >> >> *Alex Ynema** *| IT Consultant >> alex at opensystems.net.au >> >> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >> Office: +61 8 9427 2500 >> Mobile: +61 404 796 894 >> >> IT Consultant for Open Systems Support >> www.opensystems.net.au >> >> >> On 29 May 2013 23:22, Michael Collins wrote: >> >>> It sounds like you have an ESL program that is not consuming events >>> quickly enough and the event queue is filling up. >>> -MC >>> >>> >>> On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: >>> >>>> so setting the limit at 150 was fine but as soon as I set it to 200 >>>> I've now hit a problem. >>>> Freeswitch has slowly grown to 372 sessions and getting lots of these >>>> errors in the cli >>>> >>>> 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any >>>> more events..... >>>> >>>> UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, >>>> 606 microseconds >>>> FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >>>> 1500 session(s) since startup >>>> 372 session(s) - 0 out of max 10 per sec >>>> 10000 session(s) max >>>> min idle cpu 0.00/100.00 >>>> Current Stack Size/Max 240K/8192K >>>> >>>> >>>> >>>> *Alex Ynema** *| IT Consultant >>>> alex at opensystems.net.au >>>> >>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>> Office: +61 8 9427 2500 >>>> Mobile: +61 404 796 894 >>>> >>>> IT Consultant for Open Systems Support >>>> www.opensystems.net.au >>>> >>>> >>>> On 29 May 2013 15:01, jay binks wrote: >>>> >>>>> http://wiki.freeswitch.org/wiki/Limit >>>>> >>>>> >>>>> >>>>> >>>>> On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >>>>> >>>>>> >>>>>> Sorry for the thread hijack, but on the other hand, is it possible to >>>>>> limit the number of outgoing CPS? Don't seem to see that either in the wiki >>>>>> or a way to make it work. >>>>>> >>>>>> -- >>>>>> Thanks, >>>>>> Muhammad Naseer Bhatti >>>>>> >>>>>> >>>>>> >>>>>> Alex Ynema wrote: >>>>>> >>>>>> Cheers Avi I've now changed that to hash as I don't need it to be >>>>>> persistent. >>>>>> What should I see in the clie to confirm this is working without >>>>>> attempting 150+ calls >>>>>> >>>>>> Basically I've added this to my default.xml >>>>>> >>>>>> >>>>> /> >>>>>> >>>>> data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>>>>> >>>>>> >>>>>> >>>>> expression="zetta-cisco-1"> >>>>>> >>>>>> >>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> expression="zetta-cisco-2"> >>>>>> >>>>>> >>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> *Alex Ynema** *| IT Consultant >>>>>> alex at opensystems.net.au >>>>>> >>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>> Office: +61 8 9427 2500 >>>>>> Mobile: +61 404 796 894 >>>>>> >>>>>> IT Consultant for Open Systems Support >>>>>> www.opensystems.net.au >>>>>> >>>>>> >>>>>> On 29 May 2013 14:13, Avi Marcus wrote: >>>>>> >>>>>>> ... just note that's stored in a database (db) not ram (hash) so if >>>>>>> you don't need to share it / have persistence, just store it in ram. >>>>>>> >>>>>>> -Avi >>>>>>> >>>>>>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema >>>>>> > wrote: >>>>>>> >>>>>>>> I've implemented this in default.xml hoping to limit each of my two >>>>>>>> gateways to 150. >>>>>>>> Based on what's in >>>>>>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>>>>>> hopefully that works. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> expression="zetta-cisco-1"> >>>>>>>> >>>>>>>> >>>>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> expression="zetta-cisco-2"> >>>>>>>> >>>>>>>> >>>>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> *Alex Ynema** *| IT Consultant >>>>>>>> alex at opensystems.net.au >>>>>>>> >>>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>> Office: +61 8 9427 2500 >>>>>>>> Mobile: +61 404 796 894 >>>>>>>> >>>>>>>> IT Consultant for Open Systems Support >>>>>>>> www.opensystems.net.au >>>>>>>> >>>>>>>> >>>>>>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>>>>>> >>>>>>>>> Hi, >>>>>>>>> I'm trying to figure out how to limit the number of calls a >>>>>>>>> Gateway is allowed to use. Our Sip provider has provided up with 200 which >>>>>>>>> I need to set within the system somehow. >>>>>>>>> What's the best way to handle it for an outgoing only system. >>>>>>>>> I've been trying to figure out how how to configure 'Rate >>>>>>>>> limiting concurrent outgoing calls via a provider' which is >>>>>>>>> mentioned in the wiki a bit but nothing specific on what to actually do. >>>>>>>>> >>>>>>>>> *Alex Ynema** *| IT Consultant >>>>>>>>> alex at opensystems.net.au >>>>>>>>> >>>>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>> Office: +61 8 9427 2500 >>>>>>>>> Mobile: +61 404 796 894 >>>>>>>>> >>>>>>>>> IT Consultant for Open Systems Support >>>>>>>>> www.opensystems.net.au >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.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 >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.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 >>>>>>> >>>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Sincerely >>>>> >>>>> Jay >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130530/54c12951/attachment-0001.html From a.afzali2003 at gmail.com Thu May 30 13:54:05 2013 From: a.afzali2003 at gmail.com (afshin afzali) Date: Thu, 30 May 2013 14:24:05 +0430 Subject: [Freeswitch-users] Encrypted RFC2833 DTMF In-Reply-To: References: <2620FFA7-1362-4255-9FA2-1BAD96AFBDEE@jerris.com> Message-ID: Sure, Thanks Brian On Wed, May 29, 2013 at 6:27 PM, Brian Foster wrote: > Your operator is likely talking about the DTMF 'method' (inband, SIP INFO, > rfc2833), not encryption. In which case yoi are probably set up to do that > already. Post a console log of a sample call on pastebin.freeswitch.orgif yiu want is to confirm, and reply to this thread with a link. > > - BDF > On May 29, 2013 9:45 AM, "afshin afzali" wrote: > >> Hi Mike, >> >> Actually I'm not. This is a request from my local operator. As you said I >> told them that is possible to encrypt entire rtp stream by SRTP / ZRTP >> techniques. She says me that (on Huawei Switch) the encryption option just >> there when she selects rfc2833 for digit handling and not on entire media >> stream !!! >> Thank you so much, >> >> Afshin >> >> >> On Wed, May 29, 2013 at 4:44 PM, Michael Jerris wrote: >> >>> We support SRTP, which encrypts the entire media stream including the >>> 2833 digits. I'm not familiar with a standard that just encrypts the dtmf. >>> Did they provide any details of what this standard is? >>> >>> Mike >>> >>> On May 29, 2013, at 5:14 AM, afshin afzali >>> wrote: >>> >>> > Hi Guys, >>> > >>> > I don't know if FreeSWITCH is capable of to exchange encrypted rfc2833 >>> dtmf digits. My voip provider uses a Huawei switch and does ask me to >>> enable dtmf encryption but I could not find such parameter in >>> sofia.xml.conf file. >>> > >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130530/cceb61c6/attachment.html From ivan at myrvold.org Thu May 30 16:41:47 2013 From: ivan at myrvold.org (Ivan C Myrvold) Date: Thu, 30 May 2013 14:41:47 +0200 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION for SPA942 Message-ID: When I call from my Bria sip phone, or incoming from my SIP provider (they use Asterisk), I always get INCOMPATIBLE_DESTINATION. See a trace here: http://pastebin.freeswitch.org/21005 This is to a Linksys SPA942. If I call between two SPA942 phones, the call gets through. Do anyone have any clue to why I get INCOMPATIBLE_DESTINATION? Ivan From kaiser at abx.de Thu May 30 14:51:46 2013 From: kaiser at abx.de (mkmk) Date: Thu, 30 May 2013 03:51:46 -0700 (PDT) Subject: [Freeswitch-users] Incoming call and Fax Problems Message-ID: <1369911106810-7591275.post@n2.nabble.com> Hi at all,a little Problem with some callers (5% of all calls).The last Number of the extension comes very late (15-30 seconds later ).The most callers Hang up.logfilehttp://pastebin.freeswitch.org/21003If I send a Fax the following text in the log.http://pastebin.freeswitch.org/21004Can you help me.mkmk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Incoming-call-and-Fax-Problems-tp7591275.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/90ae3b87/attachment.html From mike at jerris.com Thu May 30 16:57:19 2013 From: mike at jerris.com (Michael Jerris) Date: Thu, 30 May 2013 08:57:19 -0400 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION for SPA942 In-Reply-To: References: Message-ID: <0C815C0B-CFFD-457B-8314-B0E616D94980@jerris.com> Debug log might help On May 30, 2013, at 8:41 AM, Ivan C Myrvold wrote: > When I call from my Bria sip phone, or incoming from my SIP provider (they use Asterisk), I always get INCOMPATIBLE_DESTINATION. > > See a trace here: > http://pastebin.freeswitch.org/21005 > > This is to a Linksys SPA942. > > If I call between two SPA942 phones, the call gets through. > > Do anyone have any clue to why I get INCOMPATIBLE_DESTINATION? From kris at kriskinc.com Thu May 30 17:53:02 2013 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 30 May 2013 09:53:02 -0400 Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION for SPA942 In-Reply-To: References: Message-ID: Classic problem. Check your ptime settings on the Linksys: http://wiki.freeswitch.org/wiki/Interop_List#Linksys_Products On Thu, May 30, 2013 at 8:41 AM, Ivan C Myrvold wrote: > When I call from my Bria sip phone, or incoming from my SIP provider (they use Asterisk), I always get INCOMPATIBLE_DESTINATION. > > See a trace here: > http://pastebin.freeswitch.org/21005 > > This is to a Linksys SPA942. > > If I call between two SPA942 phones, the call gets through. > > Do anyone have any clue to why I get INCOMPATIBLE_DESTINATION? > > Ivan > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Kristian Kielhofner From B.Tietz at pinguin.ag Thu May 30 18:08:13 2013 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Thu, 30 May 2013 16:08:13 +0200 Subject: [Freeswitch-users] Errer In-Reply-To: References: <07BF4904977CC645B485E970424193AD130923A407@localhost> Message-ID: <07BF4904977CC645B485E970424193AD130923A48E@localhost> Hi MC, it seems like the 1.2.9 from the archives is doing t38_gateway perfectly so something between 1.2.9 (archived one.. I suppose you now which git-version that is) and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the t38_gateway failing. regards, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Mittwoch, 29. Mai 2013 17:30 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Errer Sounds like you'll need to open a Jira on this. If you can isolate which version broke it, or ideally, which commit broke it then that would be stellar. Are you familiar with git bisect? That would assist with the troubleshooting process. -MC On Wed, May 29, 2013 at 7:41 AM, > wrote: Hi, I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... FS 1.2.5.3 works! Dialplan is Anything I can try?! VG, Benjamin T. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/e158e810/attachment-0001.html From ivan at myrvold.org Thu May 30 18:53:29 2013 From: ivan at myrvold.org (imyrvold) Date: Thu, 30 May 2013 07:53:29 -0700 (PDT) Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION for SPA942 In-Reply-To: References: Message-ID: <1369925609778-7591280.post@n2.nabble.com> That was indeed the case. Thank you so much! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/INCOMPATIBLE-DESTINATION-for-SPA942-tp7591276p7591280.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu May 30 20:34:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 30 May 2013 09:34:51 -0700 Subject: [Freeswitch-users] How can I cancel a play_and_get_digits? In-Reply-To: <2745495820513918199@unknownmsgid> References: <20130529234134.9d4e7dd4@mail.tritonwest.net> <2745495820513918199@unknownmsgid> Message-ID: Yay! Did you get the dead tree version or the electrons-only version? -MC On Wed, May 29, 2013 at 6:38 PM, Guillermo Ruiz Camauer wrote: > I already bought mine and read several chapters! I particularly liked the > one on security. > > Guillermo > > Sent from my iPhone > > On 29/05/2013, at 20:44, "Dave R. Kompel" wrote: > > You can't, at least from what I can tell. Look at the source code for it > in mod_dptools. If you need this control over it you may want to implment > DTMF buffers in your ESL app, from DTMF events, and do it all your self. > The othe option you have is to write your own DialPaln app (scripting > language or something) that can comunicate out of band via events to your > ESL program, that you use for DTMF input. > > What is your application? Maybe ESL isn't the right tool for the job. It > seems like for some reason, everyone jumps to ESL as the only recomended > and supported way of doing things external. It is NOT. The new book that > should be available does a better job of clearing that up. > > Can't wait for the new book to be released!!!! > > --Dave > > ------------------------------ > *From:* D D [mailto:tru083 at yahoo.com ] > *To:* freeswitch-users at lists.freeswitch.org [ > mailto:freeswitch-users at lists.freeswitch.org > ] > *Sent:* Wed, 29 May 2013 16:13:50 -0700 > *Subject:* [Freeswitch-users] How can I cancel a play_and_get_digits? > > Hi, > > I am using an play_and_get_digits in an ESL program. > > Based on an external event, I need to cancel the play_and_get_digits > operation. > If I send a "break", it only seems to stop the audio from playing, but the > play_and_get_digits operation continues, and later the response event > comes. > > How can I stop the entire play_and_get_digits operation from ESL? > > Thanks, > David > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/85c6646d/attachment.html From grcamauer at gmail.com Thu May 30 21:21:17 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Thu, 30 May 2013 14:21:17 -0300 Subject: [Freeswitch-users] How can I cancel a play_and_get_digits? In-Reply-To: References: <20130529234134.9d4e7dd4@mail.tritonwest.net> <2745495820513918199@unknownmsgid> Message-ID: Electronic, PDF in fact. Guillermo On Thu, May 30, 2013 at 1:34 PM, Michael Collins wrote: > Yay! Did you get the dead tree version or the electrons-only version? > -MC > > > On Wed, May 29, 2013 at 6:38 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I already bought mine and read several chapters! I particularly liked >> the one on security. >> >> Guillermo >> >> Sent from my iPhone >> >> On 29/05/2013, at 20:44, "Dave R. Kompel" wrote: >> >> You can't, at least from what I can tell. Look at the source code for >> it in mod_dptools. If you need this control over it you may want to >> implment DTMF buffers in your ESL app, from DTMF events, and do it all your >> self. The othe option you have is to write your own DialPaln app (scripting >> language or something) that can comunicate out of band via events to your >> ESL program, that you use for DTMF input. >> >> What is your application? Maybe ESL isn't the right tool for the job. It >> seems like for some reason, everyone jumps to ESL as the only recomended >> and supported way of doing things external. It is NOT. The new book that >> should be available does a better job of clearing that up. >> >> Can't wait for the new book to be released!!!! >> >> --Dave >> >> ------------------------------ >> *From:* D D [mailto:tru083 at yahoo.com ] >> *To:* freeswitch-users at lists.freeswitch.org [ >> mailto:freeswitch-users at lists.freeswitch.org >> ] >> *Sent:* Wed, 29 May 2013 16:13:50 -0700 >> *Subject:* [Freeswitch-users] How can I cancel a play_and_get_digits? >> >> Hi, >> >> I am using an play_and_get_digits in an ESL program. >> >> Based on an external event, I need to cancel the play_and_get_digits >> operation. >> If I send a "break", it only seems to stop the audio from playing, but >> the >> play_and_get_digits operation continues, and later the response event >> comes. >> >> How can I stop the entire play_and_get_digits operation from ESL? >> >> Thanks, >> David >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/fa40c7ca/attachment-0001.html From mike at hendrienet.com Fri May 31 01:16:21 2013 From: mike at hendrienet.com (Mike Hendrie) Date: Thu, 30 May 2013 16:16:21 -0500 Subject: [Freeswitch-users] Struggling Here Message-ID: Cannot get the SIP phone to ring... Where am I supposed to add the route? In the dialplan, but what am I doing wrong? 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No Route, Aborting Registrations: ================================================================================================= Call-ID: 3624a118-f19b051d-a4d9fed2 at 10.2.1.50 User: 1000 at gothamcity.xom Contact: "user" Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Status: Registered(TCP-NAT)(unknown) EXP(2013-05-30 08:13:09) EXPSECS(179) Host: GothamCity-00 IP: 10.2.1.50 Port: 33228 Auth-User: 1000 Auth-Realm: gothamcity.xom MWI-Account: 1000 at gothamcity.xom Total items returned: 1 ================================================================================================= /dialplan/public/GothamCity-00.xom.xml =========================== fs_cli when calling the number: EXECUTE sofia/external/7777777777 at 66.66.66.66 transfer(1000 GothamCity.xom.xml default) 2013-05-30 08:04:03.407264 [DEBUG] switch_ivr.c:1815 (sofia/external/ 7777777777 at 66.66.66.66) State Change CS_EXECUTE -> CS_ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777 at 66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:860 Send signal sofia/external/7777777777 at 66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [NOTICE] switch_ivr.c:1822 Transfer sofia/external/7777777777 at 66.66.66.66 to GothamCity.xom.xml[1000 at default] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:477 (sofia/external/7777777777 at 66.66.66.66) State EXECUTE going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 (sofia/external/7777777777 at 66.66.66.66) State ROUTING 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:137 sofia/external/ 7777777777 at 66.66.66.66 SOFIA ROUTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:117 sofia/external/7777777777 at 66.66.66.66 Standard ROUTING 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No Route, Aborting 2013-05-30 08:04:03.407264 [NOTICE] switch_core_state_machine.c:193 Hangup sofia/external/7777777777 at 66.66.66.66 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2013-05-30 08:04:03.407264 [DEBUG] switch_channel.c:3096 Send signal sofia/external/7777777777 at 66.66.66.66 [KILL] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777 at 66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 (sofia/external/7777777777 at 66.66.66.66) State ROUTING going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_HANGUP 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 (sofia/external/7777777777 at 66.66.66.66) State HANGUP 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:463 Channel sofia/external/ 7777777777 at 66.66.66.66 hanging up, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:597 Responding to INVITE with: 404 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:48 sofia/external/7777777777 at 66.66.66.66 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 (sofia/external/7777777777 at 66.66.66.66) State HANGUP going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:689 (sofia/external/7777777777 at 66.66.66.66) Callstate Change RINGING -> HANGUP 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:446 (sofia/external/7777777777 at 66.66.66.66) State Change CS_HANGUP -> CS_REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777 at 66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 (sofia/external/7777777777 at 66.66.66.66) State REPORTING 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:92 sofia/external/7777777777 at 66.66.66.66 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 (sofia/external/7777777777 at 66.66.66.66) State REPORTING going to sleep 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:440 (sofia/external/7777777777 at 66.66.66.66) State Change CS_REPORTING -> CS_DESTROY 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal sofia/external/7777777777 at 66.66.66.66 [BREAK] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1548 Session 1 (sofia/external/7777777777 at 66.66.66.66) Locked, Waiting on external entities 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1566 Session 1 (sofia/external/7777777777 at 66.66.66.66) Ended 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1570 Close Channel sofia/external/7777777777 at 66.66.66.66 [CS_DESTROY] 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777777777 at 66.66.66.66) Callstate Change HANGUP -> DOWN 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:568 (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_DESTROY 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:578 (sofia/external/7777777777 at 66.66.66.66) State DESTROY 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:373 sofia/external/ 7777777777 at 66.66.66.66 SOFIA DESTROY -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/4d4a441b/attachment.html From ratner2 at gmail.com Fri May 31 01:46:18 2013 From: ratner2 at gmail.com (bratner bratner) Date: Fri, 31 May 2013 00:46:18 +0300 Subject: [Freeswitch-users] Struggling Here In-Reply-To: References: Message-ID: I don't think you should specify the name of the XML file just the type of the dialplan: XML make sure it is included in context default and it has extension 1000 On Fri, May 31, 2013 at 12:16 AM, Mike Hendrie wrote: > Cannot get the SIP phone to ring... > > Where am I supposed to add the route? In the dialplan, but what am I doing > wrong? > > 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No > Route, Aborting > > Registrations: > > ================================================================================================= > Call-ID: 3624a118-f19b051d-a4d9fed2 at 10.2.1.50 > User: 1000 at gothamcity.xom > Contact: "user" ;transport=tcp;fs_nat=yes;fs_path=sip%3A1000%4010.2.1.50%3A33228%3Btransport%3Dtcp> > Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 > Status: Registered(TCP-NAT)(unknown) EXP(2013-05-30 08:13:09) > EXPSECS(179) > Host: GothamCity-00 > IP: 10.2.1.50 > Port: 33228 > Auth-User: 1000 > Auth-Realm: gothamcity.xom > MWI-Account: 1000 at gothamcity.xom > > Total items returned: 1 > > ================================================================================================= > > /dialplan/public/GothamCity-00.xom.xml > > > > > > > > > > > > =========================== > > fs_cli when calling the number: > > EXECUTE sofia/external/7777777777 at 66.66.66.66 transfer(1000 > GothamCity.xom.xml default) > 2013-05-30 08:04:03.407264 [DEBUG] switch_ivr.c:1815 (sofia/external/ > 7777777777 at 66.66.66.66) State Change CS_EXECUTE -> CS_ROUTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/7777777777 at 66.66.66.66 [BREAK] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:860 Send signal > sofia/external/7777777777 at 66.66.66.66 [BREAK] > 2013-05-30 08:04:03.407264 [NOTICE] switch_ivr.c:1822 Transfer > sofia/external/7777777777 at 66.66.66.66 to GothamCity.xom.xml[1000 at default] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:477 > (sofia/external/7777777777 at 66.66.66.66) State EXECUTE going to sleep > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_ROUTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/7777777777 at 66.66.66.66) State ROUTING > 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:137 sofia/external/ > 7777777777 at 66.66.66.66 SOFIA ROUTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:117 > sofia/external/7777777777 at 66.66.66.66 Standard ROUTING > 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No > Route, Aborting > 2013-05-30 08:04:03.407264 [NOTICE] switch_core_state_machine.c:193 Hangup > sofia/external/7777777777 at 66.66.66.66 [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2013-05-30 08:04:03.407264 [DEBUG] switch_channel.c:3096 Send signal > sofia/external/7777777777 at 66.66.66.66 [KILL] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/7777777777 at 66.66.66.66 [BREAK] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 > (sofia/external/7777777777 at 66.66.66.66) State ROUTING going to sleep > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_HANGUP > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/7777777777 at 66.66.66.66) State HANGUP > 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:463 Channel sofia/external/ > 7777777777 at 66.66.66.66 hanging up, cause: NO_ROUTE_DESTINATION > 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:597 Responding to INVITE > with: 404 > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:48 > sofia/external/7777777777 at 66.66.66.66 Standard HANGUP, cause: > NO_ROUTE_DESTINATION > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 > (sofia/external/7777777777 at 66.66.66.66) State HANGUP going to sleep > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:689 > (sofia/external/7777777777 at 66.66.66.66) Callstate Change RINGING -> HANGUP > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:446 > (sofia/external/7777777777 at 66.66.66.66) State Change CS_HANGUP -> > CS_REPORTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/7777777777 at 66.66.66.66 [BREAK] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 > (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_REPORTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/7777777777 at 66.66.66.66) State REPORTING > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:92 > sofia/external/7777777777 at 66.66.66.66 Standard REPORTING, cause: > NO_ROUTE_DESTINATION > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 > (sofia/external/7777777777 at 66.66.66.66) State REPORTING going to sleep > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:440 > (sofia/external/7777777777 at 66.66.66.66) State Change CS_REPORTING -> > CS_DESTROY > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal > sofia/external/7777777777 at 66.66.66.66 [BREAK] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1548 Session 1 > (sofia/external/7777777777 at 66.66.66.66) Locked, Waiting on external > entities > 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1566 Session 1 > (sofia/external/7777777777 at 66.66.66.66) Ended > 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1570 Close > Channel sofia/external/7777777777 at 66.66.66.66 [CS_DESTROY] > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:565 > (sofia/external/7777777777 at 66.66.66.66) Callstate Change HANGUP -> DOWN > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:568 > (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_DESTROY > 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:578 > (sofia/external/7777777777 at 66.66.66.66) State DESTROY > 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:373 sofia/external/ > 7777777777 at 66.66.66.66 SOFIA DESTROY > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/c9d99410/attachment-0001.html From ratner2 at gmail.com Fri May 31 01:50:44 2013 From: ratner2 at gmail.com (bratner bratner) Date: Fri, 31 May 2013 00:50:44 +0300 Subject: [Freeswitch-users] Struggling Here In-Reply-To: References: Message-ID: Just to make my thoughts clearer: GothamCity.xom.xml has something like On Fri, May 31, 2013 at 12:46 AM, bratner bratner wrote: > I don't think you should specify the name of the XML file > just the type of the dialplan: XML > make sure it is included in context default and it has extension 1000 > > > On Fri, May 31, 2013 at 12:16 AM, Mike Hendrie wrote: > >> Cannot get the SIP phone to ring... >> >> Where am I supposed to add the route? In the dialplan, but what am I >> doing wrong? >> >> 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No >> Route, Aborting >> >> Registrations: >> >> ================================================================================================= >> Call-ID: 3624a118-f19b051d-a4d9fed2 at 10.2.1.50 >> User: 1000 at gothamcity.xom >> Contact: "user" > ;transport=tcp;fs_nat=yes;fs_path=sip%3A1000%4010.2.1.50%3A33228%3Btransport%3Dtcp> >> Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 >> Status: Registered(TCP-NAT)(unknown) EXP(2013-05-30 08:13:09) >> EXPSECS(179) >> Host: GothamCity-00 >> IP: 10.2.1.50 >> Port: 33228 >> Auth-User: 1000 >> Auth-Realm: gothamcity.xom >> MWI-Account: 1000 at gothamcity.xom >> >> Total items returned: 1 >> >> ================================================================================================= >> >> /dialplan/public/GothamCity-00.xom.xml >> >> >> >> >> >> >> >> >> >> >> >> =========================== >> >> fs_cli when calling the number: >> >> EXECUTE sofia/external/7777777777 at 66.66.66.66 transfer(1000 >> GothamCity.xom.xml default) >> 2013-05-30 08:04:03.407264 [DEBUG] switch_ivr.c:1815 (sofia/external/ >> 7777777777 at 66.66.66.66) State Change CS_EXECUTE -> CS_ROUTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal >> sofia/external/7777777777 at 66.66.66.66 [BREAK] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:860 Send signal >> sofia/external/7777777777 at 66.66.66.66 [BREAK] >> 2013-05-30 08:04:03.407264 [NOTICE] switch_ivr.c:1822 Transfer >> sofia/external/7777777777 at 66.66.66.66 to GothamCity.xom.xml[1000 at default] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:477 >> (sofia/external/7777777777 at 66.66.66.66) State EXECUTE going to sleep >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_ROUTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/7777777777 at 66.66.66.66) State ROUTING >> 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:137 sofia/external/ >> 7777777777 at 66.66.66.66 SOFIA ROUTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:117 >> sofia/external/7777777777 at 66.66.66.66 Standard ROUTING >> 2013-05-30 08:04:03.407264 [INFO] switch_core_state_machine.c:192 No >> Route, Aborting >> 2013-05-30 08:04:03.407264 [NOTICE] switch_core_state_machine.c:193 >> Hangup sofia/external/7777777777 at 66.66.66.66 [CS_ROUTING] >> [NO_ROUTE_DESTINATION] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_channel.c:3096 Send signal >> sofia/external/7777777777 at 66.66.66.66 [KILL] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal >> sofia/external/7777777777 at 66.66.66.66 [BREAK] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:470 >> (sofia/external/7777777777 at 66.66.66.66) State ROUTING going to sleep >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_HANGUP >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/7777777777 at 66.66.66.66) State HANGUP >> 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:463 Channel sofia/external/ >> 7777777777 at 66.66.66.66 hanging up, cause: NO_ROUTE_DESTINATION >> 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:597 Responding to INVITE >> with: 404 >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:48 >> sofia/external/7777777777 at 66.66.66.66 Standard HANGUP, cause: >> NO_ROUTE_DESTINATION >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:676 >> (sofia/external/7777777777 at 66.66.66.66) State HANGUP going to sleep >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:689 >> (sofia/external/7777777777 at 66.66.66.66) Callstate Change RINGING -> >> HANGUP >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:446 >> (sofia/external/7777777777 at 66.66.66.66) State Change CS_HANGUP -> >> CS_REPORTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal >> sofia/external/7777777777 at 66.66.66.66 [BREAK] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:415 >> (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_REPORTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/7777777777 at 66.66.66.66) State REPORTING >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:92 >> sofia/external/7777777777 at 66.66.66.66 Standard REPORTING, cause: >> NO_ROUTE_DESTINATION >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:761 >> (sofia/external/7777777777 at 66.66.66.66) State REPORTING going to sleep >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:440 >> (sofia/external/7777777777 at 66.66.66.66) State Change CS_REPORTING -> >> CS_DESTROY >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1340 Send signal >> sofia/external/7777777777 at 66.66.66.66 [BREAK] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_session.c:1548 Session 1 >> (sofia/external/7777777777 at 66.66.66.66) Locked, Waiting on external >> entities >> 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1566 Session 1 >> (sofia/external/7777777777 at 66.66.66.66) Ended >> 2013-05-30 08:04:03.407264 [NOTICE] switch_core_session.c:1570 Close >> Channel sofia/external/7777777777 at 66.66.66.66 [CS_DESTROY] >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:565 >> (sofia/external/7777777777 at 66.66.66.66) Callstate Change HANGUP -> DOWN >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:568 >> (sofia/external/7777777777 at 66.66.66.66) Running State Change CS_DESTROY >> 2013-05-30 08:04:03.407264 [DEBUG] switch_core_state_machine.c:578 >> (sofia/external/7777777777 at 66.66.66.66) State DESTROY >> 2013-05-30 08:04:03.407264 [DEBUG] mod_sofia.c:373 sofia/external/ >> 7777777777 at 66.66.66.66 SOFIA DESTROY >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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/20130531/ee3e1295/attachment.html From jpyle at fidelityvoice.com Fri May 31 01:56:44 2013 From: jpyle at fidelityvoice.com (Jeff Pyle) Date: Thu, 30 May 2013 17:56:44 -0400 Subject: [Freeswitch-users] inherit_codec not working with renegotiate-codec-on-reinvite Message-ID: <51A7CB1C.5040900@fidelityvoice.com> Hello, This is on version 1.2.10 from the apt repository. I have a functioning configuration with late negotiation and inherit_codec. If I enable renegotiate-codec-on-reinvite in the relevant sofia profiles, the A-leg no longer receives the codec of the B-leg on the initial INVITE setup. Is this expected? I see relevant discussion in FS-3739 but I'm having trouble wrapping my head around whether it relates to this situation. - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/fd505cc0/attachment.html From nneul at mst.edu Fri May 31 03:03:50 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Thu, 30 May 2013 18:03:50 -0500 Subject: [Freeswitch-users] Book! Message-ID: <51A7DAD6.3040300@mst.edu> Just showed up today, and looks great! -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From msc at freeswitch.org Fri May 31 03:17:51 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 30 May 2013 16:17:51 -0700 Subject: [Freeswitch-users] Book! In-Reply-To: <51A7DAD6.3040300@mst.edu> References: <51A7DAD6.3040300@mst.edu> Message-ID: w00t! -MC On Thu, May 30, 2013 at 4:03 PM, Nathan Neulinger wrote: > Just showed up today, and looks great! > > -- Nathan > > ------------------------------------------------------------ > Nathan Neulinger nneul at mst.edu > Missouri S&T Information Technology (573) 612-1412 > System Administrator - Architect > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/88548b7a/attachment.html From msc at freeswitch.org Fri May 31 03:20:38 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 30 May 2013 16:20:38 -0700 Subject: [Freeswitch-users] Gateway Call Limits In-Reply-To: References: <51A5A5AD.1010605@gmail.com> Message-ID: Talk to Areski. He built it and I'm sure he would love to know about any limitations or bugs that you may have found. They have a git-hub repo w/ the requisite issue tracker: https://github.com/Star2Billing/newfies-dialer/issues -MC On Wed, May 29, 2013 at 11:00 PM, Alex Ynema wrote: > Okay so it's a fault with newfies rather than a problem with Freeswitch. > Any suggestions on where I should look to work out what exactly is > happening/not happening with those Event Sessions not being cleared out > from the process. > > > *Alex Ynema** *| IT Consultant > alex at opensystems.net.au > > Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 > Office: +61 8 9427 2500 > Mobile: +61 404 796 894 > > IT Consultant for Open Systems Support > www.opensystems.net.au > > > On 30 May 2013 10:25, Brian Foster wrote: > >> Newfies Dialer uses the ESL. See >> http://www.newfies-dialer.org/documentation/how-it-works/ >> >> - BDF >> On May 29, 2013 10:22 PM, "Alex Ynema" wrote: >> >>> What do you mean by ESL Program. This system is running a newfies-dialer >>> frontend with a pretty stock install of Freeswitch. >>> >>> >>> *Alex Ynema** *| IT Consultant >>> alex at opensystems.net.au >>> >>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>> Office: +61 8 9427 2500 >>> Mobile: +61 404 796 894 >>> >>> IT Consultant for Open Systems Support >>> www.opensystems.net.au >>> >>> >>> On 29 May 2013 23:22, Michael Collins wrote: >>> >>>> It sounds like you have an ESL program that is not consuming events >>>> quickly enough and the event queue is filling up. >>>> -MC >>>> >>>> >>>> On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: >>>> >>>>> so setting the limit at 150 was fine but as soon as I set it to 200 >>>>> I've now hit a problem. >>>>> Freeswitch has slowly grown to 372 sessions and getting lots of these >>>>> errors in the cli >>>>> >>>>> 2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any >>>>> more events..... >>>>> >>>>> UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, >>>>> 606 microseconds >>>>> FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >>>>> 1500 session(s) since startup >>>>> 372 session(s) - 0 out of max 10 per sec >>>>> 10000 session(s) max >>>>> min idle cpu 0.00/100.00 >>>>> Current Stack Size/Max 240K/8192K >>>>> >>>>> >>>>> >>>>> *Alex Ynema** *| IT Consultant >>>>> alex at opensystems.net.au >>>>> >>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>> Office: +61 8 9427 2500 >>>>> Mobile: +61 404 796 894 >>>>> >>>>> IT Consultant for Open Systems Support >>>>> www.opensystems.net.au >>>>> >>>>> >>>>> On 29 May 2013 15:01, jay binks wrote: >>>>> >>>>>> http://wiki.freeswitch.org/wiki/Limit >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >>>>>> >>>>>>> >>>>>>> Sorry for the thread hijack, but on the other hand, is it possible >>>>>>> to limit the number of outgoing CPS? Don't seem to see that either in the >>>>>>> wiki or a way to make it work. >>>>>>> >>>>>>> -- >>>>>>> Thanks, >>>>>>> Muhammad Naseer Bhatti >>>>>>> >>>>>>> >>>>>>> >>>>>>> Alex Ynema wrote: >>>>>>> >>>>>>> Cheers Avi I've now changed that to hash as I don't need it to be >>>>>>> persistent. >>>>>>> What should I see in the clie to confirm this is working without >>>>>>> attempting 150+ calls >>>>>>> >>>>>>> Basically I've added this to my default.xml >>>>>>> >>>>>>> >>>>>> /> >>>>>>> >>>>>> data="loopback/context/zetta-cisco-1,loopback/context/zetta-cisco-2" /> >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="zetta-cisco-1"> >>>>>>> >>>>>>> >>>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="zetta-cisco-2"> >>>>>>> >>>>>>> >>>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> *Alex Ynema** *| IT Consultant >>>>>>> alex at opensystems.net.au >>>>>>> >>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>> Office: +61 8 9427 2500 >>>>>>> Mobile: +61 404 796 894 >>>>>>> >>>>>>> IT Consultant for Open Systems Support >>>>>>> www.opensystems.net.au >>>>>>> >>>>>>> >>>>>>> On 29 May 2013 14:13, Avi Marcus wrote: >>>>>>> >>>>>>>> ... just note that's stored in a database (db) not ram (hash) so if >>>>>>>> you don't need to share it / have persistence, just store it in ram. >>>>>>>> >>>>>>>> -Avi >>>>>>>> >>>>>>>> On Wed, May 29, 2013 at 9:07 AM, Alex Ynema < >>>>>>>> alex at opensystems.net.au> wrote: >>>>>>>> >>>>>>>>> I've implemented this in default.xml hoping to limit each of my >>>>>>>>> two gateways to 150. >>>>>>>>> Based on what's in >>>>>>>>> http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits so >>>>>>>>> hopefully that works. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> expression="zetta-cisco-1"> >>>>>>>>> >>>>>>>>> >>>>>>>> data="sofia/gateway/zetta-cisco-1/${destnum}" /> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> expression="zetta-cisco-2"> >>>>>>>>> >>>>>>>>> >>>>>>>> data="sofia/gateway/zetta-cisco-2/${destnum}" /> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> *Alex Ynema** *| IT Consultant >>>>>>>>> alex at opensystems.net.au >>>>>>>>> >>>>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>> Office: +61 8 9427 2500 >>>>>>>>> Mobile: +61 404 796 894 >>>>>>>>> >>>>>>>>> IT Consultant for Open Systems Support >>>>>>>>> www.opensystems.net.au >>>>>>>>> >>>>>>>>> >>>>>>>>> On 29 May 2013 13:59, Alex Ynema wrote: >>>>>>>>> >>>>>>>>>> Hi, >>>>>>>>>> I'm trying to figure out how to limit the number of calls a >>>>>>>>>> Gateway is allowed to use. Our Sip provider has provided up with 200 which >>>>>>>>>> I need to set within the system somehow. >>>>>>>>>> What's the best way to handle it for an outgoing only system. >>>>>>>>>> I've been trying to figure out how how to configure 'Rate >>>>>>>>>> limiting concurrent outgoing calls via a provider' which is >>>>>>>>>> mentioned in the wiki a bit but nothing specific on what to actually do. >>>>>>>>>> >>>>>>>>>> *Alex Ynema** *| IT Consultant >>>>>>>>>> alex at opensystems.net.au >>>>>>>>>> >>>>>>>>>> Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>>> Office: +61 8 9427 2500 >>>>>>>>>> Mobile: +61 404 796 894 >>>>>>>>>> >>>>>>>>>> IT Consultant for Open Systems Support >>>>>>>>>> www.opensystems.net.au >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at freeswitch.org >>>>>>>> http://www.freeswitchsolutions.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 >>>>>>>> >>>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.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://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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Sincerely >>>>>> >>>>>> Jay >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/72e14166/attachment-0001.html From msc at freeswitch.org Fri May 31 03:23:12 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 30 May 2013 16:23:12 -0700 Subject: [Freeswitch-users] Incoming call and Fax Problems In-Reply-To: <1369911106810-7591275.post@n2.nabble.com> References: <1369911106810-7591275.post@n2.nabble.com> Message-ID: Have you asked the caller why there's this delay? Seems to me that this is the real issue. -MC On Thu, May 30, 2013 at 3:51 AM, mkmk wrote: > Hi at all, a little Problem with some callers (5% of all calls). The last > Number of the extension comes very late (15-30 seconds later ). The most > callers Hang up. logfile http://pastebin.freeswitch.org/21003 If I send a > Fax the following text in the log. http://pastebin.freeswitch.org/21004Can you help me. mkmk > ------------------------------ > View this message in context: Incoming call and Fax Problems > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/4707b977/attachment.html From msc at freeswitch.org Fri May 31 03:24:00 2013 From: msc at freeswitch.org (Michael Collins) Date: Thu, 30 May 2013 16:24:00 -0700 Subject: [Freeswitch-users] Errer In-Reply-To: <07BF4904977CC645B485E970424193AD130923A48E@localhost> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> Message-ID: Excellent! Now put that in a new ticket at jira.freeswitch.org and the devs will check it out. -MC On Thu, May 30, 2013 at 7:08 AM, wrote: > Hi MC,**** > > ** ** > > it seems like the 1.2.9 from the archives is doing t38_gateway perfectly** > ** > > ** ** > > so something between 1.2.9 (archived one.. I suppose you now which > git-version that is)**** > > and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the t38_gateway > failing.**** > > ** ** > > regards,**** > > Benjamin T.**** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Michael > Collins > *Gesendet:* Mittwoch, 29. Mai 2013 17:30 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Errer**** > > ** ** > > Sounds like you'll need to open a Jira on this. If you can isolate which > version broke it, or ideally, which commit broke it then that would be > stellar. Are you familiar with git bisect? That would assist with the > troubleshooting process.**** > > -MC**** > > ** ** > > On Wed, May 29, 2013 at 7:41 AM, wrote:**** > > Hi,**** > > **** > > I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X > bytes)" with FS 1.2.10 with t38_gateway. **** > > Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... > **** > > **** > > FS 1.2.5.3 works!**** > > **** > > Dialplan is**** > > **** > > **** > > **** > > **** > > **** > > ** > ** > > **** > > **** > > **** > > data="{sip_execute_on_image='t38_gateway self nocng'}sofia/intern/${ > num}@192.168.1.1:41000"/>**** > > **** > > **** > > **** > > **** > > Anything I can try?!**** > > **** > > VG,**** > > Benjamin T.**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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**** > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/537ccc0d/attachment.html From ostolyar at netflix.com Fri May 31 03:38:43 2013 From: ostolyar at netflix.com (Oleg Stolyar) Date: Thu, 30 May 2013 16:38:43 -0700 Subject: [Freeswitch-users] Can't change external RTP IP Message-ID: Hi guys, I am trying to change expternal_rtp_ip for one of my sip profiles. To make thing as simple as possible, I changed it in the internal profile (internal.xml). I just set it to a literal IP address. Then I registered a soft phone on the FS and tried to make a call. The call went through and FS console said that it connected to vociemail but I could not hear anything. Wireshark said that the SDP address that FS sent back is still FS's internal IP address. Any ideas what we are doing wrong? Thank you *Oleg* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130530/781cb47c/attachment.html From william.king at quentustech.com Fri May 31 09:07:42 2013 From: william.king at quentustech.com (William King) Date: Thu, 30 May 2013 22:07:42 -0700 Subject: [Freeswitch-users] Better approach for multi-tenant SIP testing (part 1) In-Reply-To: References: Message-ID: <51A8301E.4000408@quentustech.com> Cal, I took a different approach and took advantage of a rack of Polycom phones and a bit of electronic tinkering. http://imgur.com/a/9JfcR I took the Polycom 550s and, using the built-in XML Command Push and the state polling features, plus some perl automation, I was able to automate a large batch of call scenarios. Then when I found that SLA and BLF lights weren't available, I assimilated the phones borg style. In the album, I'm using an Arduino board, a muxer(to talk to multiple chips easily), and a set of light sensors(color, and intensity) to now detect presence issues in FreeSWITCH. 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 05/28/2013 07:03 AM, Cal Leeming [Simplicity Media Ltd] wrote: > Hello all, > > After nearly a year of active development on building our own voice > platform, we've adopted some cool tricks to speed up the workflow and > make testing less tedious. > > Over the next few months I'll be posting more articles on the tricks we > have used and the lessons we have learnt. > > Here is part 1 - a write up about a neat hack we used to virtualize > phone testing; > http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing > > For those that don't have time to read, here is a screenshot; > http://i.imgur.com/e9kUdxg.jpg > > This trick, which we've used in development for approx 6 months, has > saved countless hours in development overheads, with one-click > start/stop functionality and all the necessary modules right there in > one place. > > Of course, this should never be used as a full replacement as nothing > beats a proper hardware test to catch those weird edge cases, but it > does allow you to do the majority of your testing (95%) from a single > screen rather than a desk full of phones. It also makes attaching > wireshark a lot easier, without having to use span ports and the alike. > > Hope this helps someone else! > > Cal > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From william.king at quentustech.com Fri May 31 09:09:24 2013 From: william.king at quentustech.com (William King) Date: Thu, 30 May 2013 22:09:24 -0700 Subject: [Freeswitch-users] no ESL event for SIP SIMPLE message if destination number prefixed by + In-Reply-To: References: <51A6729C.1020404@quentustech.com> Message-ID: <51A83084.6050303@quentustech.com> Ok, did you file a Jira? I'll take a look tomorrow and see if I can replicate the issue. 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 05/29/2013 10:41 PM, Anton VG wrote: > I left an empty extension in chat plan, matching (.*). But I do not > relay on chat plan, I connect ESL INBOUND or do /events plain all in > fs_cli and send messages the same way. There is just no any events if i > prefix the number with "+". Regardless, I have tried with \+?(.*) > condition in chat plan and synthetic reply. Does not work anyway. > > > 2013/5/30 William King > > > Does your chat plan attempt to match with the + prefix? > > 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 05/29/2013 01:23 PM, Anton VG wrote: > > Hi! > > > > Just was playing with SIP SIMPLE ESL messages, actually all works fine > > if message sent to 1234567890 at freeswitch.host. > > BUT if message is sent to +1234567890 at freeswitch.host (+ prefix!)- no > > event fired :( I can see the PCAP the message accepted by FS the same > > way. So + somehow forces fs to not fire the message. Did anyone > > encountered such behavior, so should I fill the bug report on JIRA? > > > > Regards, > > Anton > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From B.Tietz at pinguin.ag Fri May 31 10:40:48 2013 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 31 May 2013 08:40:48 +0200 Subject: [Freeswitch-users] Errer In-Reply-To: References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> Message-ID: <07BF4904977CC645B485E970424193AD130923A4A1@localhost> Hi, I just created a jira-ticket, but the 1.2.10 version was not selectable, so I chose 1.2.9 VG, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Freitag, 31. Mai 2013 01:24 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Errer Excellent! Now put that in a new ticket at jira.freeswitch.org and the devs will check it out. -MC On Thu, May 30, 2013 at 7:08 AM, > wrote: Hi MC, it seems like the 1.2.9 from the archives is doing t38_gateway perfectly so something between 1.2.9 (archived one.. I suppose you now which git-version that is) and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the t38_gateway failing. regards, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Mittwoch, 29. Mai 2013 17:30 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Errer Sounds like you'll need to open a Jira on this. If you can isolate which version broke it, or ideally, which commit broke it then that would be stellar. Are you familiar with git bisect? That would assist with the troubleshooting process. -MC On Wed, May 29, 2013 at 7:41 AM, > wrote: Hi, I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... FS 1.2.5.3 works! Dialplan is Anything I can try?! VG, Benjamin T. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/9e3dee68/attachment.html From william.king at quentustech.com Fri May 31 12:01:49 2013 From: william.king at quentustech.com (William King) Date: Fri, 31 May 2013 01:01:49 -0700 Subject: [Freeswitch-users] Errer In-Reply-To: <07BF4904977CC645B485E970424193AD130923A4A1@localhost> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> <07BF4904977CC645B485E970424193AD130923A4A1@localhost> Message-ID: <51A858ED.9070603@quentustech.com> If you have access to test with the source would you be able to run a git bisect? This link should help(if you know the working commit, and a broken one you can start at step 3): http://webchick.net/node/99 Another good resource for running a git bisect: http://ruturaj.net/git-bisect-tutorial/ 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 05/30/2013 11:40 PM, B.Tietz at pinguin.ag wrote: > Hi, > > > > I just created a jira-ticket, but the 1.2.10 version was not selectable, > so I chose 1.2.9 > > > > VG, > > Benjamin T. > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Michael Collins > *Gesendet:* Freitag, 31. Mai 2013 01:24 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Errer > > > > Excellent! Now put that in a new ticket at jira.freeswitch.org > and the devs will check it out. > > -MC > > > > On Thu, May 30, 2013 at 7:08 AM, > wrote: > > Hi MC, > > > > it seems like the 1.2.9 from the archives is doing t38_gateway perfectly > > > > so something between 1.2.9 (archived one.. I suppose you now which > git-version that is) > > and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the > t38_gateway failing. > > > > regards, > > Benjamin T. > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *Im Auftrag > von *Michael Collins > *Gesendet:* Mittwoch, 29. Mai 2013 17:30 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Errer > > > > Sounds like you'll need to open a Jira on this. If you can isolate > which version broke it, or ideally, which commit broke it then that > would be stellar. Are you familiar with git bisect? That would > assist with the troubleshooting process. > > -MC > > > > On Wed, May 29, 2013 at 7:41 AM, > wrote: > > Hi, > > > > I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding > UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. > > Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 > rxfax ... > > > > FS 1.2.5.3 works! > > > > Dialplan is > > > > > > > > expression="^(01234.+)$"> > > > > data="codec_string=PCMA,PCMU,G729"/> > > > > > > > > data="{sip_execute_on_image='t38_gateway self > nocng'}sofia/intern/${num}@192.168.1.1:41000"/> > > > > > > > > > > Anything I can try?! > > > > VG, > > Benjamin T. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > From rnbrady at gmail.com Fri May 31 12:19:53 2013 From: rnbrady at gmail.com (Richard Brady) Date: Fri, 31 May 2013 09:19:53 +0100 Subject: [Freeswitch-users] sip_append_audio_sdp In-Reply-To: References: Message-ID: Bump ;-) On 20 May 2013 16:43, Richard Brady wrote: > Hi folks > > I am having some trouble with the sip_append_audio_sdp variable. If I set > it like: > > > > It seems to have absolutely no effect. I'm pretty sure I have seen it > working in the past, so now I'm wondering what the prerequisites are for it > to work (i.e. what am I doing wrong?!). > > I am using G729 in pass-through mode and have: > > inbound-late-negotiation [true] > inbound-codec-negotiation [generous] > > The expected behaviour is for the value of this variable to be appended to > the local SDP on the B-leg. The actual behaviour is that nothing is > appended. > > Any thoughts? > > Richard > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/7eda544c/attachment.html From GB at cm.nl Fri May 31 12:21:11 2013 From: GB at cm.nl (Grant Bagdasarian) Date: Fri, 31 May 2013 10:21:11 +0200 Subject: [Freeswitch-users] Integration of OpenVXI and Oktopous Message-ID: Hello, Does anyone have experience with integrating OpenVXI and Oktopous ccXML Browser with FreeSwitch? Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/34df355b/attachment.html From B.Tietz at pinguin.ag Fri May 31 12:29:37 2013 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 31 May 2013 10:29:37 +0200 Subject: [Freeswitch-users] Errer In-Reply-To: <51A858ED.9070603@quentustech.com> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> <07BF4904977CC645B485E970424193AD130923A4A1@localhost> <51A858ED.9070603@quentustech.com> Message-ID: <07BF4904977CC645B485E970424193AD130923A4BE@localhost> OK. I'll try to bisect Then I have to get the git number from the "archived" 1.2.9 It's not mentioned in fs_cli -x 'status' :-/ regards, Benjamin T. -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von William King Gesendet: Freitag, 31. Mai 2013 10:02 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] Errer If you have access to test with the source would you be able to run a git bisect? This link should help(if you know the working commit, and a broken one you can start at step 3): http://webchick.net/node/99 Another good resource for running a git bisect: http://ruturaj.net/git-bisect-tutorial/ 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 05/30/2013 11:40 PM, B.Tietz at pinguin.ag wrote: > Hi, > > > > I just created a jira-ticket, but the 1.2.10 version was not > selectable, so I chose 1.2.9 > > > > VG, > > Benjamin T. > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Michael Collins > *Gesendet:* Freitag, 31. Mai 2013 01:24 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Errer > > > > Excellent! Now put that in a new ticket at jira.freeswitch.org > and the devs will check it out. > > -MC > > > > On Thu, May 30, 2013 at 7:08 AM, > wrote: > > Hi MC, > > > > it seems like the 1.2.9 from the archives is doing t38_gateway > perfectly > > > > so something between 1.2.9 (archived one.. I suppose you now which > git-version that is) > > and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the > t38_gateway failing. > > > > regards, > > Benjamin T. > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *Im Auftrag > von *Michael Collins > *Gesendet:* Mittwoch, 29. Mai 2013 17:30 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] Errer > > > > Sounds like you'll need to open a Jira on this. If you can isolate > which version broke it, or ideally, which commit broke it then that > would be stellar. Are you familiar with git bisect? That would > assist with the troubleshooting process. > > -MC > > > > On Wed, May 29, 2013 at 7:41 AM, > wrote: > > Hi, > > > > I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding > UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. > > Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 > rxfax ... > > > > FS 1.2.5.3 works! > > > > Dialplan is > > > > > > > > expression="^(01234.+)$"> > > > > data="codec_string=PCMA,PCMU,G729"/> > > data="continue_on_fail=true"/> > > data="t38_passthru=true"/> > > data="fax_enable_t38=true"/> > > data="{sip_execute_on_image='t38_gateway self > nocng'}sofia/intern/${num}@192.168.1.1:41000"/> > > > > > > > > > > Anything I can try?! > > > > VG, > > Benjamin T. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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-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://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 From peter at olssononline.se Fri May 31 12:41:37 2013 From: peter at olssononline.se (Peter Olsson) Date: Fri, 31 May 2013 10:41:37 +0200 Subject: [Freeswitch-users] Errer In-Reply-To: <07BF4904977CC645B485E970424193AD130923A4BE@localhost> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> <07BF4904977CC645B485E970424193AD130923A4A1@localhost> <51A858ED.9070603@quentustech.com> <07BF4904977CC645B485E970424193AD130923A4BE@localhost> Message-ID: You can probably find the git revision if you check what revision the 1.2.9 tag in git was created from. 2013/5/31 > OK. I'll try to bisect > > Then I have to get the git number from the "archived" 1.2.9 > It's not mentioned in fs_cli -x 'status' :-/ > > regards, > Benjamin T. > > -----Urspr?ngliche Nachricht----- > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von William King > Gesendet: Freitag, 31. Mai 2013 10:02 > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Errer > > If you have access to test with the source would you be able to run a git > bisect? > > This link should help(if you know the working commit, and a broken one you > can start at step 3): > http://webchick.net/node/99 > > Another good resource for running a git bisect: > http://ruturaj.net/git-bisect-tutorial/ > > > 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 05/30/2013 11:40 PM, B.Tietz at pinguin.ag wrote: > > Hi, > > > > > > > > I just created a jira-ticket, but the 1.2.10 version was not > > selectable, so I chose 1.2.9 > > > > > > > > VG, > > > > Benjamin T. > > > > > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > > *Michael Collins > > *Gesendet:* Freitag, 31. Mai 2013 01:24 > > *An:* FreeSWITCH Users Help > > *Betreff:* Re: [Freeswitch-users] Errer > > > > > > > > Excellent! Now put that in a new ticket at jira.freeswitch.org > > and the devs will check it out. > > > > -MC > > > > > > > > On Thu, May 30, 2013 at 7:08 AM, > > wrote: > > > > Hi MC, > > > > > > > > it seems like the 1.2.9 from the archives is doing t38_gateway > > perfectly > > > > > > > > so something between 1.2.9 (archived one.. I suppose you now which > > git-version that is) > > > > and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the > > t38_gateway failing. > > > > > > > > regards, > > > > Benjamin T. > > > > > > > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > > ] *Im Auftrag > > von *Michael Collins > > *Gesendet:* Mittwoch, 29. Mai 2013 17:30 > > *An:* FreeSWITCH Users Help > > *Betreff:* Re: [Freeswitch-users] Errer > > > > > > > > Sounds like you'll need to open a Jira on this. If you can isolate > > which version broke it, or ideally, which commit broke it then that > > would be stellar. Are you familiar with git bisect? That would > > assist with the troubleshooting process. > > > > -MC > > > > > > > > On Wed, May 29, 2013 at 7:41 AM, > > wrote: > > > > Hi, > > > > > > > > I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding > > UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. > > > > Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 > > rxfax ... > > > > > > > > FS 1.2.5.3 works! > > > > > > > > Dialplan is > > > > > > > > > > > > > > > > > expression="^(01234.+)$"> > > > > > > > > > data="codec_string=PCMA,PCMU,G729"/> > > > > > data="continue_on_fail=true"/> > > > > > data="t38_passthru=true"/> > > > > > data="fax_enable_t38=true"/> > > > > > data="{sip_execute_on_image='t38_gateway self > > nocng'}sofia/intern/${num}@192.168.1.1:41000"/> > > > > > > > > > > > > > > > > > > > > Anything I can try?! > > > > > > > > VG, > > > > Benjamin T. > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > > > > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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-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://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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/ea77c67e/attachment-0001.html From ashish at nms.co.in Fri May 31 13:19:17 2013 From: ashish at nms.co.in (Ashish gautam) Date: Fri, 31 May 2013 14:49:17 +0530 Subject: [Freeswitch-users] Originate status Message-ID: I have the following scenario: 1. A command line php script sends a dialstring to event socket to originate call to PSTN network (Through PRI interface) 2. When the call is answered, a perl script gets executed which plays an IVR to the callee. 3. Call is hungup when the script completes Now I want to know the status of origination of the call .i.e. whether the call was successfully generated, failed, or any error/problem in origination. How can I get this status? uuid dump doesn't work when the call is not generated for some reason because the session never gets initiated. Also Event socket returns '+OK' but that specifies that the command has been received properly and has been processed tell me if I am wrong. Please throw some light Thanks. -- Ashish Gautam Nucleus Microsystems (Pvt.) Ltd. Ph. 011 47574758 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/fdd6b878/attachment.html From cal.leeming at simplicitymedialtd.co.uk Fri May 31 13:27:06 2013 From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd]) Date: Fri, 31 May 2013 10:27:06 +0100 Subject: [Freeswitch-users] Better approach for multi-tenant SIP testing (part 1) In-Reply-To: <51A8301E.4000408@quentustech.com> References: <51A8301E.4000408@quentustech.com> Message-ID: Lmao - that is pretty freaking awesome, thanks for sharing! This certainly takes automated unit testing to a whole new level, and would be great for blackbox tests on real phones - it would be interesting to see this approach on multiple different phone models as well. Cal On Fri, May 31, 2013 at 6:07 AM, William King wrote: > Cal, > > I took a different approach and took advantage of a rack of Polycom > phones and a bit of electronic tinkering. > > http://imgur.com/a/9JfcR > > I took the Polycom 550s and, using the built-in XML Command Push and the > state polling features, plus some perl automation, I was able to > automate a large batch of call scenarios. Then when I found that SLA and > BLF lights weren't available, I assimilated the phones borg style. > > In the album, I'm using an Arduino board, a muxer(to talk to multiple > chips easily), and a set of light sensors(color, and intensity) to now > detect presence issues in FreeSWITCH. > > 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 05/28/2013 07:03 AM, Cal Leeming [Simplicity Media Ltd] wrote: > > Hello all, > > > > After nearly a year of active development on building our own voice > > platform, we've adopted some cool tricks to speed up the workflow and > > make testing less tedious. > > > > Over the next few months I'll be posting more articles on the tricks we > > have used and the lessons we have learnt. > > > > Here is part 1 - a write up about a neat hack we used to virtualize > > phone testing; > > > http://blog.simplicitymedialtd.co.uk/533/quick-and-easy-approach-for-multi-tenant-sip-testing > > > > For those that don't have time to read, here is a screenshot; > > http://i.imgur.com/e9kUdxg.jpg > > > > This trick, which we've used in development for approx 6 months, has > > saved countless hours in development overheads, with one-click > > start/stop functionality and all the necessary modules right there in > > one place. > > > > Of course, this should never be used as a full replacement as nothing > > beats a proper hardware test to catch those weird edge cases, but it > > does allow you to do the majority of your testing (95%) from a single > > screen rather than a desk full of phones. It also makes attaching > > wireshark a lot easier, without having to use span ports and the alike. > > > > Hope this helps someone else! > > > > Cal > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/5af8cbe1/attachment.html From tculjaga at gmail.com Fri May 31 13:32:33 2013 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 31 May 2013 11:32:33 +0200 Subject: [Freeswitch-users] CHANNEL_HANGUP vs CHANNEL_HANGUP_COMPLETE In-Reply-To: <699859B0-6E1D-49BF-AA48-F808D9263E4C@jerris.com> References: <20130529190229.38fe4428@mail.tritonwest.net> <699859B0-6E1D-49BF-AA48-F808D9263E4C@jerris.com> Message-ID: thanks for the info ... so we will use CHANNEL_HANGUP instead of CHANNEL_HANGUP_COMPLETE :=) T. On Wed, May 29, 2013 at 9:56 PM, Michael Jerris wrote: > If you are writing a cdr module, you should be using the on_reporting > state handler, not the event handler for hangup complete. > > Mike > > On May 29, 2013, at 3:02 PM, Dave R. Kompel wrote: > > Also worth noting, CHANNEL_HANGUP_COMPLETE is the event you use to write > CDRs if you're going to do your own CDR module, or do it external. > > --Dave > > ------------------------------ > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Wed, 29 May 2013 11:19:02 -0700 > *Subject:* Re: [Freeswitch-users] CHANNEL_HANGUP vs > CHANNEL_HANGUP_COMPLETE > > Hangup is when it hangs up, hangup_complete is after the cdr processing is > complete right before destroy. > > > > On Tue, May 28, 2013 at 3:50 PM, Tihomir Culjaga > wrote: > >> hello >> >> >> im wondering what should we track as hangup event ... CHANNEL_HANGUP or >> CHANNEL_HANGUP_COMPLETE ... >> >> i have a situation where A calls FS, an ESL application answers this >> call, originates a new call to B and bridges A and B... than ESL originate >> another call towards C and joins all 3 channels into a conference. >> >> now , A hangs up, FS sends just CHANNEL_HANGUP event ... not >> CHANNEL_HANGUP_COMPLETE ... >> >> >> im wondering if it is supposed to be like that or we need to get >> CHANNEL_HANGUP_COMPLETE ? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/2fcd895c/attachment.html From B.Tietz at pinguin.ag Fri May 31 14:27:07 2013 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 31 May 2013 12:27:07 +0200 Subject: [Freeswitch-users] Errer In-Reply-To: References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> Message-ID: <07BF4904977CC645B485E970424193AD130923A4ED@localhost> Hi, as metioned in the jira (FS-5486) version c80d768 is bad (Error decoding UDPTL...) version bd4ea84 is good and finally version f3393ef (the one between the god and the ugly) can't be compiled: cc1: warnings being treated as errors udptl.c: In function 'decode_open_type': udptl.c:78: error: comparison between pointer and integer make[4]: *** [mod_spandsp_la-udptl.lo] Fehler 1 make[3]: *** [mod_spandsp-all] Fehler 1 make[2]: *** [all-recursive] Fehler 1 make[1]: *** [all-recursive] Fehler 1 make: *** [all] Fehler 2 regards, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Freitag, 31. Mai 2013 01:24 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Errer Excellent! Now put that in a new ticket at jira.freeswitch.org and the devs will check it out. -MC On Thu, May 30, 2013 at 7:08 AM, > wrote: Hi MC, it seems like the 1.2.9 from the archives is doing t38_gateway perfectly so something between 1.2.9 (archived one.. I suppose you now which git-version that is) and 1.2.10 git acc8eb5 (1.2.10 git e1a7734 also!) makes the t38_gateway failing. regards, Benjamin T. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Mittwoch, 29. Mai 2013 17:30 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Errer Sounds like you'll need to open a Jira on this. If you can isolate which version broke it, or ideally, which commit broke it then that would be stellar. Are you familiar with git bisect? That would assist with the troubleshooting process. -MC On Wed, May 29, 2013 at 7:41 AM, > wrote: Hi, I'm getting "mod_spandsp_fax.c:1691 sofia/... Error decoding UDPTL (X bytes)" with FS 1.2.10 with t38_gateway. Path is Carrier Alaw => FS1-Profile1 => FS1-Profile2 => T38 FS2 rxfax ... FS 1.2.5.3 works! Dialplan is Anything I can try?! VG, Benjamin T. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/8b131c0b/attachment-0001.html From ashwinrath at gmail.com Fri May 31 15:59:14 2013 From: ashwinrath at gmail.com (Ashwin Rath) Date: Fri, 31 May 2013 17:29:14 +0530 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: References: Message-ID: Thanks Anthony , Looks like we would need to purchase a G729 license for Freeswitch or use a Sangoma card. Is there some configuration which tells Freeswitch to use the card ( or the software codec ) only in case of a ptime mismatch so that we could save on the licensing fee only for calls that actually need ptime conversion ? On Tue, May 28, 2013 at 8:21 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Set the variable passthru_ptime_mismatch=true on both legs of the call. > That's the best we can do. If that doesn't work you'll need licenses. > > > > > On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath wrote: > >> Hi >> >> I am trying to configure FS as a B2BUA which can do ptime conversions >> between the ingress and egress leg. >> >> The question is, if both legs have G729 as the codec but ptime 20 on one >> and 30 on another, would i still need to get the licensed mod_com_g729 or >> can it work with the passthrough mod_g729. I already tried it out, but it >> doesnt seem to work. >> >> Is there some configuration needed for this ? >> >> -- >> Thegrid >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/a95eddb4/attachment.html From steveayre at gmail.com Fri May 31 16:25:42 2013 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 31 May 2013 13:25:42 +0100 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: References: Message-ID: First have you tried passthru_ptime_mismatch=true yet? It's not the default used by the passthrough mod_g729 codec. Only one module can be loaded at a time to provide a given codec (ie mod_g729 or mod_com_g729 or mod_sangoma_codec). Internally perhaps the sangoma module would avoid licenses, but they would need to clarify. AFAIK mod_com_g729 only uses a license when required, simple 729-729 bridges still use passthrough, but I don't know how ptime affects that. It's be easy to test though - load it without any licenses and try a 729-729 call with differing primes and see it the call gets through. -Steve On Friday, May 31, 2013, Ashwin Rath wrote: > Thanks Anthony , > > Looks like we would need to purchase a G729 license for Freeswitch or use > a Sangoma card. Is there some configuration which tells Freeswitch to use > the card ( or the software codec ) only in case of a ptime mismatch so > that we could save on the licensing fee only for calls that actually need > ptime conversion ? > > > On Tue, May 28, 2013 at 8:21 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > Set the variable passthru_ptime_mismatch=true on both legs of the call. > That's the best we can do. If that doesn't work you'll need licenses. > > > > > On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath wrote: > > Hi > > I am trying to configure FS as a B2BUA which can do ptime conversions > between the ingress and egress leg. > > The question is, if both legs have G729 as the codec but ptime 20 on one > and 30 on another, would i still need to get the licensed mod_com_g729 or > can it work with the passthrough mod_g729. I already tried it out, but it > doesnt seem to work. > > Is there some configuration needed for this ? > > -- > Thegrid > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > > > Ashwin Kumar Rath > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/f0a5aedc/attachment.html From ashwinrath at gmail.com Fri May 31 17:23:43 2013 From: ashwinrath at gmail.com (Ashwin Rath) Date: Fri, 31 May 2013 18:53:43 +0530 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: References: Message-ID: Hi Steven Yes we have tried the passthru_ptime_mismatch set to true but that didnt solve the purpose. So we decided to try sangoma. Trouble is FS uses the sangoma even in case of passthrough. Is there something that can be done to use mod_G729 for passthrough and sangoma g729 for ptime mismatch ? On Fri, May 31, 2013 at 5:55 PM, Steven Ayre wrote: > First have you tried passthru_ptime_mismatch=true yet? It's not the > default used by the passthrough mod_g729 codec. > > Only one module can be loaded at a time to provide a given codec (ie > mod_g729 or mod_com_g729 or mod_sangoma_codec). Internally perhaps the > sangoma module would avoid licenses, but they would need to clarify. AFAIK > mod_com_g729 only uses a license when required, simple 729-729 bridges > still use passthrough, but I don't know how ptime affects that. It's be > easy to test though - load it without any licenses and try a 729-729 call > with differing primes and see it the call gets through. > > -Steve > > > > On Friday, May 31, 2013, Ashwin Rath wrote: > >> Thanks Anthony , >> >> Looks like we would need to purchase a G729 license for Freeswitch or use >> a Sangoma card. Is there some configuration which tells Freeswitch to use >> the card ( or the software codec ) only in case of a ptime mismatch so >> that we could save on the licensing fee only for calls that actually need >> ptime conversion ? >> >> >> On Tue, May 28, 2013 at 8:21 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >> Set the variable passthru_ptime_mismatch=true on both legs of the call. >> That's the best we can do. If that doesn't work you'll need licenses. >> >> >> >> >> On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath wrote: >> >> Hi >> >> I am trying to configure FS as a B2BUA which can do ptime conversions >> between the ingress and egress leg. >> >> The question is, if both legs have G729 as the codec but ptime 20 on one >> and 30 on another, would i still need to get the licensed mod_com_g729 or >> can it work with the passthrough mod_g729. I already tried it out, but it >> doesnt seem to work. >> >> Is there some configuration needed for this ? >> >> -- >> Thegrid >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> >> >> Ashwin Kumar Rath >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/33d0a3e2/attachment-0001.html From steveu at coppice.org Fri May 31 18:35:26 2013 From: steveu at coppice.org (Steve Underwood) Date: Fri, 31 May 2013 22:35:26 +0800 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: References: Message-ID: <51A8B52E.4070508@coppice.org> Hi, The Sangoma G.729 is a transcoder. You don't want that for ptime resolution. Decompressing and recompressing really hurts the quality. FS will repack the frames to resolve ptime mismatches, although the latency is better if you can avoid such repacking. Steve On 05/31/2013 09:23 PM, Ashwin Rath wrote: > Hi Steven > > Yes we have tried the passthru_ptime_mismatch set to true but that > didnt solve the purpose. So we decided to try sangoma. Trouble is FS > uses the sangoma even in case of passthrough. Is there something that > can be done to use mod_G729 for passthrough and sangoma g729 for ptime > mismatch ? > > > On Fri, May 31, 2013 at 5:55 PM, Steven Ayre > wrote: > > First have you tried passthru_ptime_mismatch=true yet? It's not > the default used by the passthrough mod_g729 codec. > > Only one module can be loaded at a time to provide a given codec > (ie mod_g729 or mod_com_g729 or mod_sangoma_codec). Internally > perhaps the sangoma module would avoid licenses, but they would > need to clarify. AFAIK mod_com_g729 only uses a license when > required, simple 729-729 bridges still use passthrough, but I > don't know how ptime affects that. It's be easy to test though - > load it without any licenses and try a 729-729 call with differing > primes and see it the call gets through. > > -Steve > > > > On Friday, May 31, 2013, Ashwin Rath wrote: > > Thanks Anthony , > > Looks like we would need to purchase a G729 license for > Freeswitch or use a Sangoma card. Is there some configuration > which tells Freeswitch to use the card ( or the software codec > ) only in case of a ptime mismatch so that we could save on > the licensing fee only for calls that actually need ptime > conversion ? > > > On Tue, May 28, 2013 at 8:21 PM, Anthony Minessale > wrote: > > Set the variable passthru_ptime_mismatch=true on both legs > of the call. That's the best we can do. If that doesn't > work you'll need licenses. > > > > > On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath > wrote: > > Hi > > I am trying to configure FS as a B2BUA which can do > ptime conversions between the ingress and egress leg. > > The question is, if both legs have G729 as the codec > but ptime 20 on one and 30 on another, would i still > need to get the licensed mod_com_g729 or can it work > with the passthrough mod_g729. I already tried it out, > but it doesnt seem to work. > > Is there some configuration needed for this ? > > -- > Thegrid > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > 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 > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > > Ashwin Kumar Rath > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > -- > Ashwin Kumar Rath > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 From tru083 at yahoo.com Fri May 31 18:38:47 2013 From: tru083 at yahoo.com (D D) Date: Fri, 31 May 2013 07:38:47 -0700 (PDT) Subject: [Freeswitch-users] Question about event-lock, break, and playback with async ESL Message-ID: <1370011127.10516.YahooMailNeo@web120702.mail.ne1.yahoo.com> Hi, If I have an async ESL playback operation running, and I issue a 'break', immediately followed by a new playback, is there any chance that the second playback will be stopped by the 'break' ? In my testing I never observed something like this happening, but I was wondering if?event-lock? is really needed in this case. ?The event-lock causes other side effects that I would like to avoid if possible. Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/eec6eb53/attachment.html From mbodbg at gmx.net Fri May 31 18:59:25 2013 From: mbodbg at gmx.net (mbo) Date: Fri, 31 May 2013 16:59:25 +0200 Subject: [Freeswitch-users] Question about event-lock, break, and playback with async ESL In-Reply-To: <1370011127.10516.YahooMailNeo@web120702.mail.ne1.yahoo.com> References: <1370011127.10516.YahooMailNeo@web120702.mail.ne1.yahoo.com> Message-ID: <5A5FC41B-2AEA-4314-9C6A-B2453BD6CA7D@gmx.net> I think if you always wait until a command has been executed before you send a new command, it can't happened that you stop the wrong playback. If you pass Event-UUID in sendmsg like ? sendmsg Event-UUID: 5bf340cd-7a7e-4965-9285-95ed365ed242 call-command: ... execute-app-name: ... execute-app-arg: ... ? you will get this UUID back as Application-UUID in the CHANNEL_EXECUTE and CHANNEL_EXECUTE_COMPLETE events, so we are able to match it. So you can wait until you get CHANNEL_EXECUTE_COMPLETE before you send your second playback. See also "Correlate SendMsg reply with request in async mode" in the mailing list. - Markus Am 31.05.2013 um 16:38 schrieb D D : > Hi, > > If I have an async ESL playback operation running, and I issue a 'break', immediately followed > by a new playback, is there any chance that the second playback will be stopped by the 'break' ? > > In my testing I never observed something like this happening, but I was wondering if event-lock > is really needed in this case. The event-lock causes other side effects that I would > like to avoid if possible. > > Thanks, > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/d5a39fa4/attachment.html From ashwinrath at gmail.com Fri May 31 19:15:49 2013 From: ashwinrath at gmail.com (Ashwin Rath) Date: Fri, 31 May 2013 20:45:49 +0530 Subject: [Freeswitch-users] G729 ptime mismatch on ingress and egress legs In-Reply-To: <51A8B52E.4070508@coppice.org> References: <51A8B52E.4070508@coppice.org> Message-ID: The problem that we are facing with the mod_G729 in a back to back scenario is that FS doesnt re packetize the RTP. For example if one stream has ptime = 10 then it remains at 10 on both the A as well as B leg. The same applies to the other which may be at another ptime (say 30). What we are trying to achieve is A leg always using ptime 10 and B leg always using ptime 30. Is this possible without using any licensed software ? On Fri, May 31, 2013 at 8:05 PM, Steve Underwood wrote: > Hi, > > The Sangoma G.729 is a transcoder. You don't want that for ptime > resolution. Decompressing and recompressing really hurts the quality. FS > will repack the frames to resolve ptime mismatches, although the latency > is better if you can avoid such repacking. > > Steve > > > On 05/31/2013 09:23 PM, Ashwin Rath wrote: > > Hi Steven > > > > Yes we have tried the passthru_ptime_mismatch set to true but that > > didnt solve the purpose. So we decided to try sangoma. Trouble is FS > > uses the sangoma even in case of passthrough. Is there something that > > can be done to use mod_G729 for passthrough and sangoma g729 for ptime > > mismatch ? > > > > > > On Fri, May 31, 2013 at 5:55 PM, Steven Ayre > > wrote: > > > > First have you tried passthru_ptime_mismatch=true yet? It's not > > the default used by the passthrough mod_g729 codec. > > > > Only one module can be loaded at a time to provide a given codec > > (ie mod_g729 or mod_com_g729 or mod_sangoma_codec). Internally > > perhaps the sangoma module would avoid licenses, but they would > > need to clarify. AFAIK mod_com_g729 only uses a license when > > required, simple 729-729 bridges still use passthrough, but I > > don't know how ptime affects that. It's be easy to test though - > > load it without any licenses and try a 729-729 call with differing > > primes and see it the call gets through. > > > > -Steve > > > > > > > > On Friday, May 31, 2013, Ashwin Rath wrote: > > > > Thanks Anthony , > > > > Looks like we would need to purchase a G729 license for > > Freeswitch or use a Sangoma card. Is there some configuration > > which tells Freeswitch to use the card ( or the software codec > > ) only in case of a ptime mismatch so that we could save on > > the licensing fee only for calls that actually need ptime > > conversion ? > > > > > > On Tue, May 28, 2013 at 8:21 PM, Anthony Minessale > > wrote: > > > > Set the variable passthru_ptime_mismatch=true on both legs > > of the call. That's the best we can do. If that doesn't > > work you'll need licenses. > > > > > > > > > > On Tue, May 28, 2013 at 9:12 AM, Ashwin Rath > > wrote: > > > > Hi > > > > I am trying to configure FS as a B2BUA which can do > > ptime conversions between the ingress and egress leg. > > > > The question is, if both legs have G729 as the codec > > but ptime 20 on one and 30 on another, would i still > > need to get the licensed mod_com_g729 or can it work > > with the passthrough mod_g729. I already tried it out, > > but it doesnt seem to work. > > > > Is there some configuration needed for this ? > > > > -- > > Thegrid > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > > Server > > > > > > 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 > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > > > Ashwin Kumar Rath > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > > > > > > > > -- > > Ashwin Kumar Rath > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > -- Ashwin Kumar Rath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/b54aca79/attachment-0001.html From alex at digitalmail.com Fri May 31 19:24:12 2013 From: alex at digitalmail.com (Alex Lake) Date: Fri, 31 May 2013 16:24:12 +0100 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: References: <51387356.7010706@digitalmail.com> <51389DF4.4090808@digitalmail.com> Message-ID: <51A8C09C.7070904@digitalmail.com> ...and for our next trick, how do I do a callback from one PSTN destination to another? i.e. place a call to 07775123456 and when that answers, place another one to 02012345678? Cheers, Alex > You almost got it: it's: exten [dialplan] [context] > > So you just needed to do: > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > 07775123456 xml dp0095 > ... without the &bridge. You /either/ choose an extension or do &(app). > Usage: originate |&() > [] [] [] [] [] > To quote from the wiki: > > "Here's an example of originating a call to an extension in a > different context than 'default' (required for the FreePBX which uses > context_1, context_2, etc.):" > > originate sofia/internal/2001 at foo.com 3001 xml context_3 > -Avi Marcus > BestFone > > > On Thu, Mar 7, 2013 at 4:02 PM, Alex Lake > wrote: > > Yes, I'd seen that - but I'm currently the wrong side of the "got > it" fence. However, I did this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302&bridge({origination_caller_id_number=2070602000}sofia/internal/07775123456 at pstngateway.com > ) > > and it kind of did what I wanted. > > However, what I really want to do is to simulate as closely as > possible what happens when ext 0095302 makes an outbound call to > 07775123456 from a handset - preferably using the dp0095 context > of the xml (?) dialplan. > > So I thought I'd try using the dialplan and context parameters > like this: > > originate > {origination_caller_id_name=CallBack,origination_caller_id_number=07775123456}user/0095302 > &bridge(07775123456) xml dp0095 > > But I've clearly got the wrong end of the stick! > >> There's a whole bunch of examples here: >> http://wiki.freeswitch.org/wiki/Mod_commands#originate >> >> The first arg rings first, and must be an endpoint, e.g. sofia/, >> user/. >> Once they pick up, the second arg is called. >> >> So originating to a local user or to a remote endpoint is nearly >> the same... especially if you can use the lcr/ endpoint. >> Your leg B can be a brige, a conference >> &conference(conf_uuid-TEST_CON), or just hit the dialplan. >> >> >> -Avi Marcus >> >> On Thu, Mar 7, 2013 at 1:00 PM, Alex Lake > > wrote: >> >> I was wondering where's a good place to find some examples of >> how, by >> sending the right commands to the event_socket, I could have >> Freeswitch >> establish callbacks for me. >> >> Essentially there are a couple of different termination types >> - PSTN >> (via a gateway) and internally registered SIP accounts >> >> I would wish to be able to set up A->B (and maybe small >> conferences) >> using these types of destination in any combination. >> >> I've looked through the event_socket pages and the >> "originate" syntax, >> but would like to put together something a little more >> "idiot-friendly" >> so am looking around for precedents/tips... >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.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 >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: >> 03/06/13 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2240 / Virus Database: 2641/5652 - Release Date: 03/06/13 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/5bb91c13/attachment.html From peter at olssononline.se Fri May 31 20:09:05 2013 From: peter at olssononline.se (Peter Olsson) Date: Fri, 31 May 2013 18:09:05 +0200 Subject: [Freeswitch-users] Question about event-lock, break, and playback with async ESL In-Reply-To: <1370011127.10516.YahooMailNeo@web120702.mail.ne1.yahoo.com> References: <1370011127.10516.YahooMailNeo@web120702.mail.ne1.yahoo.com> Message-ID: You shouldn't need event-lock in this case, as long you don't try to queue up any more playback events while playing a file. And yes, an immediate playback after a break will always work, but you will need to execute break the same way you execute playback - to make sure they end up in the same queue. If you do that it will work fine, and it's always safe that an immediate playback after the break will be executed as expected. /Peter 2013/5/31 D D > Hi, > > If I have an async ESL playback operation running, and I issue a 'break', > immediately followed > by a new playback, is there any chance that the second playback will be > stopped by the 'break' ? > > In my testing I never observed something like this happening, but I was > wondering if event-lock > is really needed in this case. The event-lock causes other side effects > that I would > like to avoid if possible. > > Thanks, > David > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/2ffa64b0/attachment-0001.html From kaiser at abx.de Fri May 31 09:52:36 2013 From: kaiser at abx.de (mkmk) Date: Thu, 30 May 2013 22:52:36 -0700 (PDT) Subject: [Freeswitch-users] Incoming call and Fax Problems In-Reply-To: References: <1369911106810-7591275.post@n2.nabble.com> Message-ID: <1369979556279-7591295.post@n2.nabble.com> I don?t asked the caller. I have 2 identical servers they running Freeswitch. Both have the same configuration. They are only on different ISDN lines. 1 Freeswitch work perfekt and the other makes problems. mkmk -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Incoming-call-and-Fax-Problems-tp7591275p7591295.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeffkingsleykigg at yahoo.com Fri May 31 10:53:15 2013 From: jeffkingsleykigg at yahoo.com (Jeffkingsley Kigg) Date: Thu, 30 May 2013 23:53:15 -0700 (PDT) Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 83, Issue 285 In-Reply-To: References: Message-ID: <1369983195.73662.YahooMailNeo@web125101.mail.ne1.yahoo.com> ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Friday, May 31, 2013 2:21 AM Subject: FreeSWITCH-users Digest, Vol 83, Issue 285 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: Book! (Michael Collins) ? 2. Re: Gateway Call Limits (Michael Collins) w00t! -MC On Thu, May 30, 2013 at 4:03 PM, Nathan Neulinger wrote: Just showed up today, and looks great! > >-- Nathan > >------------------------------------------------------------ >Nathan Neulinger ? ? ? ? ? ? ? ? ? ? ? nneul at mst.edu >Missouri S&T Information Technology ? ?(573) 612-1412 >System Administrator - Architect > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org Talk to Areski. He built it and I'm sure he would love to know about any limitations or bugs that you may have found. They have a git-hub repo w/ the requisite issue tracker: https://github.com/Star2Billing/newfies-dialer/issues -MC On Wed, May 29, 2013 at 11:00 PM, Alex Ynema wrote: Okay so it's a fault with newfies rather than a problem with Freeswitch. Any suggestions on where I should look to work out what exactly is happening/not happening with those Event Sessions not being cleared out from the process. > > > > >Alex Ynema?|?IT Consultant >alex at opensystems.net.au > > >Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >Office: +61 8 9427 2500 >Mobile: +61 404 796 894 > > > >IT Consultant for Open Systems Support > >www.opensystems.net.au > > >On 30 May 2013 10:25, Brian Foster wrote: > >Newfies Dialer uses the ESL. See http://www.newfies-dialer.org/documentation/how-it-works/ >>- BDF >>On May 29, 2013 10:22 PM, "Alex Ynema" wrote: >> >>What do you mean by ESL Program. This system is running a newfies-dialer frontend with a pretty stock install of Freeswitch. >>> >>> >>> >>> >>>Alex Ynema?|?IT Consultant >>>alex at opensystems.net.au >>> >>> >>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>Office: +61 8 9427 2500 >>>Mobile: +61 404 796 894 >>> >>> >>> >>>IT Consultant for Open Systems Support >>> >>>www.opensystems.net.au >>> >>> >>>On 29 May 2013 23:22, Michael Collins wrote: >>> >>>It sounds like you have an ESL program that is not consuming events quickly enough and the event queue is filling up. >>>>-MC >>>> >>>> >>>> >>>> >>>>On Wed, May 29, 2013 at 12:17 AM, Alex Ynema wrote: >>>> >>>>so setting the limit at 150 was fine but as soon as I set it to 200 I've now hit a problem. >>>>>Freeswitch has slowly grown to 372 sessions and getting lots of these errors in the cli >>>>> >>>>> >>>>> >>>>>2013-05-29 15:16:41.022625 [ERR] switch_cpp.cpp:48 Cannot queue any more events..... >>>>> >>>>> >>>>>UP 0 years, 0 days, 0 hours, 23 minutes, 55 seconds, 273 milliseconds, 606 microseconds >>>>>FreeSWITCH (Version 1.5.1b git d2f3a31 2013-05-21 02:00:43Z) is ready >>>>>1500 session(s) since startup >>>>>372 session(s) - 0 out of max 10 per sec >>>>>10000 session(s) max >>>>>min idle cpu 0.00/100.00 >>>>>Current Stack Size/Max 240K/8192K >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>Alex Ynema?|?IT Consultant >>>>>alex at opensystems.net.au >>>>> >>>>> >>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>Office: +61 8 9427 2500 >>>>>Mobile: +61 404 796 894 >>>>> >>>>> >>>>> >>>>>IT Consultant for Open Systems Support >>>>> >>>>>www.opensystems.net.au >>>>> >>>>> >>>>>On 29 May 2013 15:01, jay binks wrote: >>>>> >>>>>http://wiki.freeswitch.org/wiki/Limit >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>On 29 May 2013 16:52, Muhammad Naseer Bhatti wrote: >>>>>> >>>>>> >>>>>>>Sorry for the thread hijack, but on the other hand, is it possible to limit the number of outgoing CPS? Don't seem to see that either in the wiki or a way to make it work. >>>>>>> >>>>>>> >>>>>>>-- >>>>>>>Thanks, >>>>>>>Muhammad Naseer Bhatti >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>Alex Ynema wrote: >>>>>>>Cheers Avi I've now changed that to hash as I don't need it to be persistent. >>>>>>>>What should I see in the clie to confirm this is working without attempting 150+ calls >>>>>>>> >>>>>>>> >>>>>>>>Basically I've added this to my default.xml >>>>>>>> >>>>>>>> >>>>>>>>? ? >>>>>>>>? ? >>>>>>>> >>>>>>>> >>>>>>>>? ? >>>>>>>>? ? ? ? >>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>? ? ? ? >>>>>>>>? ? >>>>>>>>? ? >>>>>>>>? ? ? ? >>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>? ? ? ? >>>>>>>>? ? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>Alex Ynema?|?IT Consultant >>>>>>>>alex at opensystems.net.au >>>>>>>> >>>>>>>> >>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>Office: +61 8 9427 2500 >>>>>>>>Mobile: +61 404 796 894 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>IT Consultant for Open Systems Support >>>>>>>> >>>>>>>>www.opensystems.net.au >>>>>>>> >>>>>>>> >>>>>>>>On 29 May 2013 14:13, Avi Marcus wrote: >>>>>>>> >>>>>>>>... just note that's stored in a database (db) not ram (hash) so if you don't need to share it / have persistence, just store it in ram. >>>>>>>>> >>>>>>>>> >>>>>>>>>-Avi >>>>>>>>> >>>>>>>>>On Wed, May 29, 2013 at 9:07 AM, Alex Ynema wrote: >>>>>>>>> >>>>>>>>>I've implemented this in default.xml hoping to limit each of my two gateways to 150.? >>>>>>>>>>Based on what's in?http://wiki.freeswitch.org/wiki/Limit#Using_limit_with_per-gateway_or_per-user_channel_limits?so hopefully that works. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>? ? >>>>>>>>>>? ? ? ? >>>>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>>>? ? ? ? >>>>>>>>>>? ? >>>>>>>>>>? ? >>>>>>>>>>? ? ? ? >>>>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>>>? ? ? ? ? ? ? ? >>>>>>>>>>? ? ? ? >>>>>>>>>>? ? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>Alex Ynema?|?IT Consultant >>>>>>>>>>alex at opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>>>Office: +61 8 9427 2500 >>>>>>>>>>Mobile: +61 404 796 894 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>IT Consultant for Open Systems Support >>>>>>>>>> >>>>>>>>>>www.opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>On 29 May 2013 13:59, Alex Ynema wrote: >>>>>>>>>> >>>>>>>>>>Hi, >>>>>>>>>>>I'm trying to figure out how to limit the number of calls a Gateway is allowed to use. Our Sip provider has provided up with 200 which I need to set within the system somehow. >>>>>>>>>>>What's the best way to handle it for an outgoing only system. >>>>>>>>>>>I've been trying to figure out how how to configure 'Rate limiting concurrent outgoing calls via a provider' which is mentioned in the wiki a bit but nothing specific on what to actually do. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>Alex Ynema?|?IT Consultant >>>>>>>>>>>alex at opensystems.net.au >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016 >>>>>>>>>>>Office: +61 8 9427 2500 >>>>>>>>>>>Mobile: +61 404 796 894 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>>IT Consultant for Open Systems Support >>>>>>>>>>> >>>>>>>>>>>www.opensystems.net.au >>>>>>>>>> >>>>>>>>>> >>>>>>>>>>_________________________________________________________________________ >>>>>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>>>>consulting at freeswitch.org >>>>>>>>>>http://www.freeswitchsolutions.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 >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>>_________________________________________________________________________ >>>>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>>>consulting at freeswitch.org >>>>>>>>>http://www.freeswitchsolutions.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 >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.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 >>>>>>>_________________________________________________________________________ >>>>>>>Professional FreeSWITCH Consulting Services: >>>>>>>consulting at freeswitch.org >>>>>>>http://www.freeswitchsolutions.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 >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>>-- >>>>>>Sincerely >>>>>> >>>>>>Jay >>>>>>_________________________________________________________________________ >>>>>>Professional FreeSWITCH Consulting Services: >>>>>>consulting at freeswitch.org >>>>>>http://www.freeswitchsolutions.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 >>>>>> >>>>>> >>>>> >>>>>_________________________________________________________________________ >>>>>Professional FreeSWITCH Consulting Services: >>>>>consulting at freeswitch.org >>>>>http://www.freeswitchsolutions.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 >>>>> >>>>> >>>> >>>> >>>>-- >>>>Michael S Collins >>>>Twitter: @mercutioviz >>>>http://www.FreeSWITCH.org >>>>http://www.ClueCon.com >>>>http://www.OSTAG.org >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.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 >>>> >>>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.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 >>> >>> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.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 >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.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 > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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/20130530/eb5b1226/attachment-0001.html From smrdoshi at gmail.com Fri May 31 11:20:35 2013 From: smrdoshi at gmail.com (smrdoshi) Date: Fri, 31 May 2013 00:20:35 -0700 (PDT) Subject: [Freeswitch-users] opensips & Freeswitch calling issue In-Reply-To: References: <1368765501780-7590795.post@n2.nabble.com> <1368781652733-7590807.post@n2.nabble.com> Message-ID: <1369984835508-7591297.post@n2.nabble.com> Hi Karsten, Thanks for the link. I got the issue. I missed the code to make feature working. I search and kept it. Now its working fine. Thanks, Samir Doshi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795p7591297.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mino.haluz at gmail.com Fri May 31 11:43:34 2013 From: mino.haluz at gmail.com (Mino Haluz) Date: Fri, 31 May 2013 09:43:34 +0200 Subject: [Freeswitch-users] Forward custom header Message-ID: Hi, I know freeswitch is b2bua, but if there is some way how could I forward custom header added earlier by proxy from one call leg to another, it would be perfect. What I'm trying to achieve in dialplan is: and bridging to gw. So this header could be transparently forwarded, but it does not add anything. Am I doing something really wrong ? Thanks, Mino -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/b1d1e0ff/attachment.html From mino.haluz at gmail.com Fri May 31 11:53:25 2013 From: mino.haluz at gmail.com (Mino Haluz) Date: Fri, 31 May 2013 09:53:25 +0200 Subject: [Freeswitch-users] Too many channels used in my scenario Message-ID: Hi, I am thinking of this topology: caller->freeswitch->proxy in proxy, there will be some routing module as carrierroute or so and then call is sent back callee<-freeswitch<-proxy thing is, if only one call is placed, 4 channels are used. If I used sangoma transcoding card, every channel has to be transcoded to the internal FS coded and back, so 8 transcoding channels in sangoma driver are used for just one call (!). How could I tell freeswitch to do not pass SDP information to the proxy side and let it use only 2 channels as in standard caller->callee call. Thanks, Mino -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/614f90f2/attachment.html From dev.2981988 at gmail.com Fri May 31 12:04:36 2013 From: dev.2981988 at gmail.com (divyeshkamothi) Date: Fri, 31 May 2013 01:04:36 -0700 (PDT) Subject: [Freeswitch-users] Is it possbile to remove Allow, Supported, Allow-Events from header in freeswitch Message-ID: <1369987476083-7591298.post@n2.nabble.com> hi all there is requirement where i need to remove Allow, Supported, Allow-Events. from fs SIP headers. please provide me way using which i can remove. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-it-possbile-to-remove-Allow-Supported-Allow-Events-from-header-in-freeswitch-tp7591298.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Fri May 31 20:33:33 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 31 May 2013 11:33:33 -0500 Subject: [Freeswitch-users] Is it possbile to remove Allow, Supported, Allow-Events from header in freeswitch In-Reply-To: <1369987476083-7591298.post@n2.nabble.com> Message-ID: I don't think you can really remove those... Is this one of those things where you have some vendor complaining because you are sending them headers they don't support so they are blaming the issues on that? On 5/31/13 3:04 AM, "divyeshkamothi" wrote: > hi all > > there is requirement where i need to remove Allow, Supported, Allow-Events. > from fs SIP headers. > > please provide me way using which i can remove. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Is-it-possbile-to-remove-Allow-S > upported-Allow-Events-from-header-in-freeswitch-tp7591298.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://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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch From ehermouet at bluetel.fr Fri May 31 20:35:16 2013 From: ehermouet at bluetel.fr (Hermouet Erwan) Date: Fri, 31 May 2013 18:35:16 +0200 Subject: [Freeswitch-users] from name and sip user In-Reply-To: References: Message-ID: <32206c71-2165-4ebd-bf6b-b13a1cce5b19@email.android.com> So nobody know ? Tks advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/322eb41c/attachment.html From krice at freeswitch.org Fri May 31 20:46:20 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 31 May 2013 11:46:20 -0500 Subject: [Freeswitch-users] Hey Guys Today 3PM EST Friday FreeForAll Message-ID: Hey Guys don?t forget to join us at 3PM EST (that?s like NOON PST) for the FridayFreeForAll For those of you Voice Only, sip:888 at conference.freeswitch.org as usual For those that want to join with Video, sip:888 at mcu.freeswitch.org For those that have the option, try the video Bridge lets see if we can find a breaking point with load! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/6dd9ac59/attachment.html From grcamauer at gmail.com Fri May 31 20:47:30 2013 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 31 May 2013 13:47:30 -0300 Subject: [Freeswitch-users] Too many channels used in my scenario In-Reply-To: References: Message-ID: Why are you counting twice? If FS just sits in the middle, would it not use just 4 channels? Is FS the only thing under your control in this scenario? Otherwise, could you not configure FS<--->Proxy to use the same ptimes so as not to use the Sangoma card? On Fri, May 31, 2013 at 4:53 AM, Mino Haluz wrote: > Hi, > > I am thinking of this topology: > > caller->freeswitch->proxy > > in proxy, there will be some routing module as carrierroute or so and then > call is sent back > > callee<-freeswitch<-proxy > > thing is, if only one call is placed, 4 channels are used. If I used > sangoma transcoding card, every channel has to be transcoded to the > internal FS coded and back, so 8 transcoding channels in sangoma driver are > used for just one call (!). How could I tell freeswitch to do not pass SDP > information to the proxy side and let it use only 2 channels as in standard > caller->callee call. > > Thanks, > Mino > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/fe6d3026/attachment.html From krice at freeswitch.org Fri May 31 21:01:39 2013 From: krice at freeswitch.org (Ken Rice) Date: Fri, 31 May 2013 12:01:39 -0500 Subject: [Freeswitch-users] Too many channels used in my scenario In-Reply-To: Message-ID: Umm FreeSWITCH can operate in bypass media mode... But I guess the real question, is why are you looping the call thru the carrier route proxy and back to FS when you can just have FS ask the outside routing where to send the call, then have FS just send the call along to the callee On 5/31/13 2:53 AM, "Mino Haluz" wrote: > Hi, > > I am thinking of this topology: > > caller->freeswitch->proxy > > in proxy, there will be some routing module as carrierroute or so and then > call is sent back > > callee<-freeswitch<-proxy > > thing is, if only one call is placed, 4 channels are used. If I used sangoma > transcoding card, every channel has to be transcoded to the internal FS coded > and back, so 8 transcoding channels in sangoma driver are used for just one > call (!). How could I tell freeswitch to do not pass SDP information to the > proxy side and let it use only 2 channels as in standard caller->callee call. > > Thanks, > Mino > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/9fb91ce3/attachment-0001.html From sdevoy at bizfocused.com Fri May 31 21:20:59 2013 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 31 May 2013 13:20:59 -0400 Subject: [Freeswitch-users] Specifying MOH Message-ID: <15b701ce5e23$38636f30$a92a4d90$@bizfocused.com> HI, I see in the wiki it says: hold_music Per-channel hold music. Supports all audio formats and audio streams. The hold_music variable can also be set globally at vars.xml. Usage: For multi-tenant environment, if you want to have a separate MOH for the phone with hold button (like Polycom) that utilizes RE-INVITE with no media ip addr (0.0.0.0) for hold, you can override the hold-music values in the sip profile parameter similar to the following example: I set this in my dial-plan like this: I had an error on the console that said "File access error." I chmod'ed the MP3 file. Now I get no error, but complete silence on hold. Any ideas? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/7b79ade9/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 70 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/7b79ade9/attachment.gif From nneul at mst.edu Fri May 31 21:29:51 2013 From: nneul at mst.edu (Nathan Neulinger) Date: Fri, 31 May 2013 12:29:51 -0500 Subject: [Freeswitch-users] Specifying MOH In-Reply-To: <15b701ce5e23$38636f30$a92a4d90$@bizfocused.com> References: <15b701ce5e23$38636f30$a92a4d90$@bizfocused.com> Message-ID: <51A8DE0F.1040907@mst.edu> Did you copy that verbatim? Cause you have "hold_music=hold_music=..." doubled. -- Nathan On 05/31/2013 12:20 PM, Sean Devoy wrote: > HI, > > I see in the wiki it says: > > > hold_music > > Per-channel hold music. Supports all audio formats and audio streams. The hold_music variable can also be set globally > at vars.xml. > > > *Usage:* > > > > For multi-tenant environment, if you want to have a separate MOH for the phone with hold button (like Polycom) that > utilizes RE-INVITE with no media ip addr (0.0.0.0) for hold, you can override the hold-music values in the sip profile > parameter similar to the following example: > > > > I set this in my dial-plan like this: > > > > I had an error on the console that said ?File access error.? I chmod?ed the MP3 file. Now I get no error, but > complete silence on hold. > > Any ideas? > > Thanks, > > Sean > -- ------------------------------------------------------------ Nathan Neulinger nneul at mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect From zoltan.medveczky at 8x8.com Fri May 31 21:30:58 2013 From: zoltan.medveczky at 8x8.com (Zoltan Medveczky) Date: Fri, 31 May 2013 10:30:58 -0700 Subject: [Freeswitch-users] Forward custom header In-Reply-To: References: Message-ID: Did you try using "export" instead of "set"? I believe you could also do something this when bridging the call: On Fri, May 31, 2013 at 12:43 AM, Mino Haluz wrote: > Hi, > > I know freeswitch is b2bua, but if there is some way how could I forward > custom header added earlier by proxy from one call leg to another, it would > be perfect. What I'm trying to achieve in dialplan is: > > > and bridging to gw. > > So this header could be transparently forwarded, but it does not add > anything. Am I doing something really wrong ? > > Thanks, > Mino > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/a0302339/attachment.html From mino.haluz at gmail.com Fri May 31 21:44:59 2013 From: mino.haluz at gmail.com (Mino Haluz) Date: Fri, 31 May 2013 19:44:59 +0200 Subject: [Freeswitch-users] Too many channels used in my scenario In-Reply-To: References: Message-ID: We would like to use FS as SBC, so accounting will be done in our core network which routes traffic among various carriers, it's ultra fast and good designed. I do not want FS to route the traffic, as with multiple endpoints synchronizing could be a problem. I will try how many G729 codecs it will consume, afterwards I can say how many channels it requires. On Fri, May 31, 2013 at 7:01 PM, Ken Rice wrote: > Umm FreeSWITCH can operate in bypass media mode... But I guess the real > question, is why are you looping the call thru the carrier route proxy and > back to FS when you can just have FS ask the outside routing where to send > the call, then have FS just send the call along to the callee > > > > > On 5/31/13 2:53 AM, "Mino Haluz" wrote: > > Hi, > > I am thinking of this topology: > > caller->freeswitch->proxy > > in proxy, there will be some routing module as carrierroute or so and then > call is sent back > > callee<-freeswitch<-proxy > > thing is, if only one call is placed, 4 channels are used. If I used > sangoma transcoding card, every channel has to be transcoded to the > internal FS coded and back, so 8 transcoding channels in sangoma driver are > used for just one call (!). How could I tell freeswitch to do not pass SDP > information to the proxy side and let it use only 2 channels as in standard > caller->callee call. > > Thanks, > Mino > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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 > > > -- > Ken > *http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > *irc.freenode.net #freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.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/20130531/8229b09b/attachment-0001.html From msc at freeswitch.org Fri May 31 23:09:40 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 31 May 2013 12:09:40 -0700 Subject: [Freeswitch-users] Errer In-Reply-To: <07BF4904977CC645B485E970424193AD130923A4ED@localhost> References: <07BF4904977CC645B485E970424193AD130923A407@localhost> <07BF4904977CC645B485E970424193AD130923A48E@localhost> <07BF4904977CC645B485E970424193AD130923A4ED@localhost> Message-ID: On Fri, May 31, 2013 at 3:27 AM, wrote: > Hi,**** > > ** ** > > as metioned in the jira (FS-5486)**** > > ** ** > > version c80d768 is bad (Error decoding UDPTL...)**** > > version bd4ea84 is good and finally**** > > version f3393ef (the one between the god and the ugly) can?t be compiled: > **** > > ** ** > > cc1: warnings being treated as errors > udptl.c: In function ?decode_open_type?: > udptl.c:78: error: comparison between pointer and integer > make[4]: *** [mod_spandsp_la-udptl.lo] Fehler 1 > make[3]: *** [mod_spandsp-all] Fehler 1 > make[2]: *** [all-recursive] Fehler 1 > make[1]: *** [all-recursive] Fehler 1 > make: *** [all] Fehler 2**** > > ** > Try "make spandsp-reconf" and then recompile. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/b17a8579/attachment.html From bdfoster at davri.com Fri May 31 23:12:51 2013 From: bdfoster at davri.com (Brian Foster) Date: Fri, 31 May 2013 15:12:51 -0400 Subject: [Freeswitch-users] sip_append_audio_sdp In-Reply-To: References: Message-ID: Post a console log of an example where this action fails (with global siptrace on) to pastebin.freeswitch.org and then post a link back to this tread. Include some context so we can better understand your log (from internal to external profiles, what phones, carriers, anything relevent and helpful). - BDF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130531/0418a24a/attachment.html From msc at freeswitch.org Fri May 31 23:15:22 2013 From: msc at freeswitch.org (Michael Collins) Date: Fri, 31 May 2013 12:15:22 -0700 Subject: [Freeswitch-users] Setting up callbacks In-Reply-To: <51A8C09C.7070904@digitalmail.com> References: <51387356.7010706@digitalmail.com> <51389DF4.4090808@digitalmail.com> <51A8C09C.7070904@digitalmail.com> Message-ID: On Fri, May 31, 2013 at 8:24 AM, Alex Lake wrote: > ...and for our next trick, how do I do a callback from one PSTN > destination to another? i.e. place a call to 07775123456 and when that > answers, place another one to 02012345678? > > Cheers, > Alex > I must be missing something because the way you phrased the question it sounds like that's exactly what the originate command does, assuming you set ignore_early_media=true. -MC -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... 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